UC540 SIP trunk "allowed IP addresses" limitation?
Testing out a UC540 and I'm using Callcentric. Working great, but the feature under the SIP trunk configuring/Advanced that allows you to set the IP addresses of machines allowed to connect to the UC540 only supports a maximum of 100 entries. Callcentric wants 204.11.192.0/24 or 204.11.192.0-204.11.192-255 unblocked.
The UC540 doesn't seem to support inputting a range in any notation I'm aware of, maybe I'm missing something?
I contacted Callcentric and asked if they could be more specific, LOL, but they said they change things on their end from time to time.
Anyone else figure out a way around this?
Okay, follow-up to the question.
How bad an idea is it to leave this security feature disabled?
I have a bit of experience in this area, I use an Asterisk (FreePBX distro) for my work phone system, and Callcentric is my SIP trunker. Before I limited incoming connections to my FreePBX server to the Callcentric range above, there would be occasions (maybe once every 2-3 weeks) where someone (probably up to no good) found my open port(s) and all my phones would start ringing simultaneously. It lasted for about 20-30 seconds, and then would stop.
My assumption was that someone was looking for security vulnerabilities so they could use my Asterisk box to place their LD (maybe even overseas) calls. They never got anywhere, though, as the Asterisk was fully patched, and my passwords for my handsets were strong.
I wonder if the Cisco UC540 will suffer the same sort of result, or worse, or maybe even not so much.
Obviously I can simply select another trunker. The thing I like about Callcentric are the rates and also the dashboard that allows me to make a lot of adjustments (like forwarding calls to a cell phone number, and the ability to configure faxing reception, too). So if anyone has a suggestion for a supported SIP trunker that supports the same types of features, I'd be open to that, as well.
BTW, I did ask Callcentric one more time, with a link to this topic, whether they can restrict that range to better than 0-255, but they declined. I see their point, just wish there was another simple way around this.
Similar Messages
-
Hi,
I´m trying to configure an YX Wireless as a SIP Trunk for the UC540. I can receive calls, but i can´t make any call, maybe someone who has already configure this can give some advice.
Any help will be appreciate.
Thanks.I think the problem is the codec. I've configured the SIP Trunk to use the codec G711ulaw, but checking the logs the outgoing calls use the codec g729r8 which is not supported by the gateway. Can someone help me change this??
Is it possible via CCA or do I have to use the CLI?? -
I've been racking my brain on this since 4am, finally went to bed, and I'm at it again.My UC is registered to voip.ms, and can place multiple outbound calls with no issue.Inbound calls were coming in intermittantly last night. Right now it just goes to an announcement: "We're sorry, but due to technical difficulties, we are unable to route your call, please try your call again"I did a debug ccsip calls and I see the calls hitting the UC, and I'm wondering if it's a NAT issue:003446: Jun 28 16:53:39.120: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr: ReqLine IP addr does not match with host IP addr003447: Jun 28 16:53:39.120: //-1/0F1EA628A9A3/SIP/Error/sact_idle_new_message_invite: Invalid URL in incoming INVITE003448: Jun 28 16:53:39.156: //-1/0F1EA628A9A3/SIP/Call/sipSPICallInfo: The Call Setup Information is:Call Control...
This topic first appeared in the Spiceworks CommunityYou would have a wan address your provider wan and your internal network
Data and VoIP could be two separate networks..
Example 192.168.1,0 for data
192.168.2.0 voice
The outbound is failing because you don't have any dial- peers pointing to your provider
For example
Dial-peer voice 1 VoIP
Destination-partners 9T
Session target ipv4: op address of sip provider
Is your sip provider just dial tone or Internet as well.
If there providing Internet you need to configure nat...
Hope is gives you a right direction
Sent from Cisco Technical Support iPad App -
Forwarding with SIP Trunking and Retaining CID
We are using SIP trunking to get to the PSTN and CUCM 7.1. The SIP provider only permits calls originating from their own DIDs.
So how can we allow our users to forward all calls to say their cell phone. AND when a call comes in we want them to be able to see the original caller ID. Is it possible? What is the mojo? Thanks!Hi,
for the SIP trunks there are no limits from the system. Check this out: https://supportforums.cisco.com/message/3795863#3795863
If you have different voice codecs for your phones and the trunk you need DSP ressources for transcoding.
As I know the UC520 have a PVDM2-64 with 4 DSP chips. You can use the DSP calculator from cisco to find out how many DSPs you need. But keep in mind that conferencing and transcoding can't share a DSP processor.
For example 1 DSP for conferencing and 3 for transcoding.
best regards
Christian -
Calls from Sip Trunk to UC540 and then to CUE returned ** Service Unavailable**
Hi to all
i have something strange here and i need your assistance
Call Flow:
Sip trunk-->UC540--> CUE
When calls coming to UC540 from outside and then going to cue then we send back service unavailable.I made a translation and i sent directly the incoming calls to CUE
The same behavior is also if i send the calls to dummy number and then from there set forward all to voice mail.
Incoming voicemail is working fine
Incoming calls to phones also ok
Uc540: 8.6
CUE: 8.6.5
A number: 99999999
B number: 22777777
Voice Mail Number:111
Attached is the trace
i see that we hit the correct dial peers .
I have enable only trancoder since MTP is not register ( don't know why , but i don't think also that is necessary..
voice service voip
ip address trusted list
ipv4 172.16.80.0 255.255.255.0
ipv4 172.16.81.0 255.255.255.0
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
sip
no update-callerid
dial-peer voice 1000 voip
description **SIP TRUNK**
translation-profile incoming SIP-INCOMING
translation-profile outgoing SIP-OUTGOING
destination-pattern 9T
modem passthrough nse codec g711alaw
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate 9600
fax protocol pass-through g711alaw
no vad
dial-peer voice 2001 voip
description ** cue voicemail pilot number **
destination-pattern 111
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number 111
no voice-class sip outbound-proxy
dtmf-relay sip-notify
codec g711ulaw
no vad
Regards
chrysostomosHi
Interface IP-Address OK? Method Status Protocol
FastEthernet0/0 unassigned YES NVRAM up up
FastEthernet0/0.10 192.168.0.10 YES DHCP up up ----> For internet
FastEthernet0/0.20 10.151.5.130 YES NVRAM up up ------> For sip trunk
In0/0 10.1.10.2 YES unset up up --------> default gw for cue
Vlan1 unassigned YES unset up up
Vlan100 unassigned YES unset up up
Vlan200 unassigned YES unset up down
Vlan300 unassigned YES unset up down
NVI0 10.1.10.2 YES unset up up
BVI1 192.168.20.1 YES NVRAM up up
BVI100 10.1.1.1 YES NVRAM up up ---------> ip for cme
Loopback0 10.1.10.2 YES NVRAM up up ------> default gw for cue
dial-peer voice 2001 voip
description ** cue voicemail pilot number **
destination-pattern 111
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number 111
no voice-class sip outbound-proxy
voice-class sip bind control source-interface BVI100
voice-class sip bind media source-interface BVI100
dtmf-relay sip-notify
codec g711ulaw
no vad
interface FastEthernet0/0.10
description **FOR INTERNET**
encapsulation dot1Q 10
ip address dhcp
ip access-group 105 in
ip nat outside
ip inspect SDM_LOW out
ip virtual-reassembly in
interface FastEthernet0/0.20
description **FOR SIP TRUNK WITH ISP**
encapsulation dot1Q 20
ip address 10.151.5.130 255.255.255.240
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
ping 10.1.10.1 source bvi100
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 10.1.10.1, timeout is 2 seconds:
Packet sent with a source address of 10.1.1.1
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/1 ms
I have bind the interface of cme ( 10.1.1.1) but the call fails again
Attached is the trace
Anything to advice? -
PMF to allow outgoing calls through SIP Trunk Without Registering
Hello,
I have an intermitant issue with one of our UC320W's running 2.3.2(6) firmware. The customers VOIP SIP trunk becomes unregistered for periods of time, stopping incoming and outgoing calls. Once unregistered it takes quite a while to rergister. Our service provider has informed us that the re-register period is the cause and we should try and shorten it, so first question is there a way to do this, also what is the re-register retry window in the first place?
I have an analogue line that can receive calls only so I have made this the fallover number with the VOIP provider, that gives a little releife for incoming calls, but not outgoing. I beleive in other phone systems a SIP trunk does not need to be registered to make an outgoing call, and it is usually an option to say only make outgoing calls if the SIP trunk is registered. I cannot find that option anywhere to deselect it, is there a PMF I could apply to allow outgoing calls without registering?
Thank you,
TonyHi Tony,
Please install the SIP_Trunk_Register_Timer.pmf at status->Devices->Alter PMFs in configure utility. Please remember to apply the configuration afterwards. This PMF can let user to select the re-register period. You can find the PMF at https://supportforums.cisco.com/docs/DOC-16301
Regards,
Wendy Yang -
Issue with instant ringback when using sip trunk to SP
Hi all,
We use CUCM 8.0.2.
We have a SIP trunk to a SP connected via one of our Cisco 2911 routers configured as a CUBE.
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.0(1)M3, RELEASE SOFTWARE (fc2)
c2900-universalk9-mz.SPA.150-1.M3.bin
Cisco CISCO2911/K9 (revision 1.0)
Technology Package License Information for Module:'c2900'
Technology Technology-package
Current Type
ipbase ipbasek9 Permanent
security securityk9 Permanent
uc uck9 Permanent
data None None
We also have several ISDN lines that run out via various Cisco routers configured as H323 gateways.
We use 7945 and CIPC for our phones.
We're having an issue with calls going via the SIP trunk where we hear ringing instantly after dialling - but before the actual device at the other end starts ringing (considerable difference).
Using the SIP trunk: If I make a call to my mobile phone - I hear ringing instantly - about 3 rings before my mobile phone actually starts ringing - undesireable.
Using H323 gateway: If I make a call to my mobile phone - I hear silence for a bit - then ringing when the mobile starts ringing - desired.
Using SIP trunk: If I make a call to a landline that is ready - it rings instantly for at least 1 ring - before the actual phone I'm calling starts ringing - undesireable.
Using H323 gateway: There is a momentary pause before hearing ringing on my phone and the phone I dialled - desired.
Using SIP trunk: If I make a call to a landline that is off-hook (with no call-waiting/etc.) - it rings once and then returns the busy signal (the worst issue) - undesireable.
Using H323 gateway: There is a momentary pause before hearing busy signal - desired.
Phone to phone internally (same network): Operates as expected (instantly rings locally and on the phone I'm calling). Between phones that utilise the SIP trunk and phones that utilise the H323 gateways within the same network - communication is instant and as expected.
Any ideas why this happens and how to stop it?
I want it to not ring until the situation is known and that it can provide the appropriate feedback (ringing/busy/etc.).
Some possibly relevant config (note that there is a known bug with this IOS that meant I had to declare the codec in each dial-peer as the voice class would not work):
voice service voip
address-hiding
mode border-element
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
header-passing error-passthru
early-offer forced
midcall-signaling passthru
interface GigabitEthernet0/0
ip address x.x.x.x 255.255.255.252
ip access-group acl.SIP-IN in
no ip redirects
no ip unreachables
ip verify unicast reverse-path
ip virtual-reassembly
duplex full
speed 100
no cdp enable
gateway
timer receive-rtp 1200
sip-ua
connection-reuse
gatekeeper
shutdown
dial-peer voice 1 voip
description *** INBOUND CALLS FROM CARRIER ***
translation-profile incoming SIPTRUNK-INCOMING
session protocol sipv2
incoming called-number #blah blah#
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 61 voip
description **** WA, SA AND NT NUMBERS ****
destination-pattern 0[8]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[8]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 81 voip
description **** MOBILE NUMBERS ****
destination-pattern 0[4]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[4]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 500 voip
description *** INBOUND SIP TRUNK TO CUCM PUB ***
translation-profile outgoing SIPTRUNK-CALLING-ADD-0
preference 1
destination-pattern 5[12]..
session protocol sipv2
session target ipv4:<OUR CUCM PUBLISHER IP>
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
Any help or a point in the right direction would be greatly appreciated.
Cheers,
BrettI ended up resolving this issue as follows:
In CUCM, under Device > Device Settings > SIP Profile.
I modifed the profile relevant to my SIP trunk, under the "Trunk Specific Configuration", I set "SIP Rel1XX Options" from "Disabled" to "Send PRACK if 1xx Contains SDP".
Now, I get the expected delay before hearing ringback.
Solved! -
SIP Trunk - No voice with Single Number Reach
Hi Community.
I setup SIP Trunk with the CCA. Everything is working Call In and Call Out. Call Forward and so on.
But with Single Number reach is something wrong. The mobile phone is ringing and I can get the call, but I hear not any voice.
Can someone please help me out? Below the config.
version 15.1
parser config cache interface
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
service internal
service compress-config
service sequence-numbers
dot11 ssid cisco-data
vlan 1
authentication open
dot11 ssid cisco-voice
vlan 100
authentication open
ip source-route
ip cef
ip dhcp relay information trust-all
ip dhcp excluded-address 10.1.1.1 10.1.1.9
ip dhcp excluded-address 10.1.1.241 10.1.1.255
ip dhcp pool phone
network 10.1.1.0 255.255.255.0
default-router 10.1.1.1
option 150 ip 10.1.1.1
ip domain name site1.365873.trk.ipvoip.ch
ip name-server 8.8.8.8
ip inspect WAAS flush-timeout 10
ip inspect name SDM_LOW dns
ip inspect name SDM_LOW ftp
ip inspect name SDM_LOW h323
ip inspect name SDM_LOW https
ip inspect name SDM_LOW icmp
ip inspect name SDM_LOW imap
ip inspect name SDM_LOW pop3
ip inspect name SDM_LOW netshow
ip inspect name SDM_LOW rcmd
ip inspect name SDM_LOW realaudio
ip inspect name SDM_LOW rtsp
ip inspect name SDM_LOW esmtp
ip inspect name SDM_LOW sqlnet
ip inspect name SDM_LOW streamworks
ip inspect name SDM_LOW tftp
ip inspect name SDM_LOW tcp router-traffic
ip inspect name SDM_LOW udp router-traffic
ip inspect name SDM_LOW vdolive
no ipv6 cef
multilink bundle-name authenticated
stcapp ccm-group 1
stcapp
isdn switch-type basic-net3
voice call send-alert
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
sip
registrar server expires max 3600 min 3600
localhost dns:site1.365873.trk.ipvoip.ch
no update-callerid
voice class codec 1
codec preference 1 g711alaw
voice register global
mode cme
source-address 10.1.1.1 port 5060
load 9971 sip9971.9-2-2
load 9951 sip9951.9-2-2
load 8961 sip8961.9-2-2
timezone 23
voice source-group CCA_SIP_SOURCE_GROUP_CUE_CME
access-list 2
translation-profile incoming SIP_Incoming
voice source-group CCA_SIP_SOURCE_GROUP_EXTERNAL
access-list 3
voice translation-rule 9
rule 1 /0041449475090/ /90/
rule 2 /0041449475091/ /91/
rule 3 /0041449475092/ /92/
rule 4 /0041449475093/ /93/
rule 5 /0041449475094/ /94/
rule 6 /0041449475095/ /95/
rule 7 /0041449475096/ /96/
rule 8 /0041449475097/ /97/
rule 9 /0041449475098/ /98/
rule 10 /0041449475099/ /99/
voice translation-rule 410
rule 1 /^0\(.*\)/ /\1/
rule 15 /^..$/ /0041449475090/
voice translation-rule 411
rule 1 /^0\(.*\)/ /ABCD0\1/
voice translation-rule 412
rule 1 /^ABCD\(.*\)/ /\1/
voice translation-rule 422
rule 15 /^ABCD\(.*\)/ /\1/
voice translation-rule 1000
rule 1 /.*/ //
voice translation-rule 1111
rule 1 /^9\([1-9]\)$/ /004144947509\1/
rule 15 /^..$/ /0041449475090/
voice translation-rule 1112
rule 1 /^0/ //
voice translation-rule 2000
rule 1 /0041449475098/ /98/
voice translation-rule 2001
rule 1 /0041449475097/ /97/
voice translation-rule 2002
rule 1 /^6/ //
voice translation-rule 2222
voice translation-profile AA_Profile
translate called 2001
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
voice translation-profile CallBlocking
translate called 2222
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
voice translation-profile PSTN_CallForwarding
translate redirect-target 410
translate redirect-called 410
voice translation-profile PSTN_Outgoing
translate calling 1111
translate called 1112
translate redirect-target 410
translate redirect-called 410
voice translation-profile SIP_Called_9
translate calling 3265
translate called 9
voice translation-profile SIP_Incoming
translate called 411
voice translation-profile SIP_Passthrough
translate called 412
voice translation-profile SIP_Passthrough_CallBlocking
translate called 422
voice translation-profile VM_Profile
translate called 2000
voice translation-profile XFER_TO_VM_PROFILE
translate redirect-called 2002
voice translation-profile nondialable
translate called 1000
voice-card 0
dspfarm
dsp services dspfarm
fax interface-type fax-mail
license udi pid UC540W-BRI-K9 sn FGL163220SL
archive
log config
logging enable
logging size 600
hidekeys
username admin privilege 15 secret xxx
username xxx password 0 ""
username xxx password 0 ""
ip tftp source-interface Loopback0
bridge irb
interface Loopback0
description $FW_INSIDE$
ip address 10.1.10.2 255.255.255.252
ip access-group 101 in
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/0
description $FW_OUTSIDE$
no ip address
ip inspect SDM_LOW out
ip virtual-reassembly in
ip verify unicast reverse-path
load-interval 30
shutdown
duplex auto
speed auto
interface Integrated-Service-Engine0/0
description cue is initialized with default IMAP group
ip unnumbered Loopback0
ip nat inside
ip virtual-reassembly in
service-module ip address 10.1.10.1 255.255.255.252
service-module ip default-gateway 10.1.10.2
interface FastEthernet0/1/0
no ip address
macro description cisco-desktop
spanning-tree portfast
interface FastEthernet0/1/1
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/2
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/3
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/4
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/5
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/6
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/7
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/8
no ip address
macro description cisco-desktop
spanning-tree portfast
interface BRI0/1/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
isdn incoming-voice voice
isdn sending-complete
isdn static-tei 0
interface BRI0/1/1
no ip address
shutdown
isdn switch-type basic-net3
isdn point-to-point-setup
isdn incoming-voice voice
isdn sending-complete
isdn static-tei 0
interface Dot11Radio0/5/0
no ip address
ssid cisco-data
ssid cisco-voice
speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0
station-role root
antenna receive right
antenna transmit right
interface Dot11Radio0/5/0.1
encapsulation dot1Q 1 native
bridge-group 1
bridge-group 1 subscriber-loop-control
bridge-group 1 spanning-disabled
bridge-group 1 block-unknown-source
no bridge-group 1 source-learning
no bridge-group 1 unicast-flooding
interface Dot11Radio0/5/0.100
encapsulation dot1Q 100
bridge-group 100
bridge-group 100 subscriber-loop-control
bridge-group 100 spanning-disabled
bridge-group 100 block-unknown-source
no bridge-group 100 source-learning
no bridge-group 100 unicast-flooding
interface Vlan1
no ip address
bridge-group 1
bridge-group 1 spanning-disabled
interface Vlan100
no ip address
bridge-group 100
bridge-group 100 spanning-disabled
interface BVI1
description $FW_INSIDE$
ip address 192.168.10.2 255.255.255.0
ip access-group 102 in
ip nat inside
ip virtual-reassembly in
interface BVI100
description $FW_INSIDE$
ip address 10.1.1.1 255.255.255.0
ip access-group 103 in
ip nat inside
ip virtual-reassembly in
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http path flash:/gui
ip dns server
ip nat inside source list 1 interface FastEthernet0/0 overload
ip route 0.0.0.0 0.0.0.0 192.168.10.1
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
access-list 1 remark SDM_ACL Category=2
access-list 1 permit 10.1.1.0 0.0.0.255
access-list 1 permit 192.168.10.0 0.0.0.255
access-list 1 permit 10.1.10.0 0.0.0.3
access-list 2 remark CCA_SIP_SOURCE_GROUP_ACL_INTERNAL
access-list 2 remark SDM_ACL Category=1
access-list 2 permit 192.168.10.2
access-list 2 permit 10.1.10.0 0.0.0.3
access-list 2 permit 192.168.10.0 0.0.0.255
access-list 2 permit 10.1.1.0 0.0.0.255
access-list 3 remark CCA_SIP_SOURCE_GROUP_ACL_EXTERNAL
access-list 3 remark SDM_ACL Category=1
access-list 3 permit 212.147.47.216
access-list 3 deny any
access-list 100 remark auto generated by SDM firewall configuration
access-list 100 remark SDM_ACL Category=1
access-list 100 deny ip 192.168.10.0 0.0.0.255 any
access-list 100 deny ip host 255.255.255.255 any
access-list 100 deny ip 127.0.0.0 0.255.255.255 any
access-list 100 permit ip any any
access-list 101 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 101 remark SDM_ACL Category=1
access-list 101 permit tcp 10.1.1.0 0.0.0.255 eq 2000 any
access-list 101 permit udp 10.1.1.0 0.0.0.255 eq 2000 any
access-list 101 deny ip 10.1.1.0 0.0.0.255 any
access-list 101 deny ip 192.168.10.0 0.0.0.255 any
access-list 101 deny ip 192.168.1.0 0.0.0.255 any
access-list 101 deny ip host 255.255.255.255 any
access-list 101 deny ip 127.0.0.0 0.255.255.255 any
access-list 101 permit ip any any
access-list 102 remark auto generated by SDM firewall configuration##NO_ACES_6##
access-list 102 remark SDM_ACL Category=1
access-list 102 deny ip 10.1.10.0 0.0.0.3 any
access-list 102 deny ip 10.1.1.0 0.0.0.255 any
access-list 102 deny ip 192.168.1.0 0.0.0.255 any
access-list 102 deny ip host 255.255.255.255 any
access-list 102 deny ip 127.0.0.0 0.255.255.255 any
access-list 102 permit ip any any
access-list 103 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 103 remark SDM_ACL Category=1
access-list 103 permit tcp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 permit udp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 deny ip 10.1.10.0 0.0.0.3 any
access-list 103 deny ip 192.168.10.0 0.0.0.255 any
access-list 103 deny ip 192.168.1.0 0.0.0.255 any
access-list 103 deny ip host 255.255.255.255 any
access-list 103 deny ip 127.0.0.0 0.255.255.255 any
access-list 103 permit ip any any
access-list 104 remark auto generated by SDM firewall configuration##NO_ACES_14##
access-list 104 remark SDM_ACL Category=1
access-list 104 deny ip 10.1.10.0 0.0.0.3 any
access-list 104 deny ip 10.1.1.0 0.0.0.255 any
access-list 104 permit ip any any
access-list 104 permit udp host 8.8.8.8 eq domain any
access-list 104 permit icmp any any echo-reply
access-list 104 permit icmp any any time-exceeded
access-list 104 permit icmp any any unreachable
access-list 104 deny ip 10.0.0.0 0.255.255.255 any
access-list 104 deny ip 172.16.0.0 0.15.255.255 any
access-list 104 deny ip 192.168.0.0 0.0.255.255 any
access-list 104 deny ip 127.0.0.0 0.255.255.255 any
access-list 104 deny ip host 255.255.255.255 any
access-list 104 deny ip host 0.0.0.0 any
access-list 104 deny ip any any
control-plane
bridge 1 route ip
bridge 100 route ip
voice-port 0/0/0
cptone CH
station-id name FAX
station-id number 99
caller-id enable
voice-port 0/0/1
cptone CH
shutdown
caller-id enable
voice-port 0/0/2
cptone CH
shutdown
caller-id enable
voice-port 0/0/3
cptone CH
shutdown
caller-id enable
voice-port 0/1/0
compand-type a-law
cptone CH
bearer-cap Speech
voice-port 0/1/1
compand-type a-law
cptone CH
bearer-cap Speech
voice-port 0/4/0
auto-cut-through
signal immediate
input gain auto-control -15
description Music On Hold Port
sccp local Loopback0
sccp ccm 10.1.1.1 identifier 1 version 4.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register mtpa4934c6ee4e0
dspfarm profile 2 transcode
description CCA transcoding for SIP Trunk VTX
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 10
associate application SCCP
dial-peer cor custom
name internal
name local
name local-plus
name international
name national
name national-plus
name emergency
name toll-free
dial-peer cor list call-internal
member internal
dial-peer cor list call-local
member local
dial-peer cor list call-local-plus
member local-plus
dial-peer cor list call-national
member national
dial-peer cor list call-national-plus
member national-plus
dial-peer cor list call-international
member international
dial-peer cor list call-emergency
member emergency
dial-peer cor list call-toll-free
member toll-free
dial-peer cor list user-internal
member internal
member emergency
dial-peer cor list user-local
member internal
member local
member emergency
member toll-free
dial-peer cor list user-local-plus
member internal
member local
member local-plus
member emergency
member toll-free
dial-peer cor list user-national
member internal
member local
member local-plus
member national
member emergency
member toll-free
dial-peer cor list user-national-plus
member internal
member local
member local-plus
member national
member national-plus
member emergency
member toll-free
dial-peer cor list user-international
member internal
member local
member local-plus
member international
member national
member national-plus
member emergency
member toll-free
dial-peer voice 1 pots
destination-pattern 99
port 0/0/0
no sip-register
dial-peer voice 2 pots
port 0/0/1
no sip-register
dial-peer voice 3 pots
port 0/0/2
no sip-register
dial-peer voice 4 pots
port 0/0/3
no sip-register
dial-peer voice 5 pots
description ** MOH Port **
destination-pattern ABC
port 0/4/0
no sip-register
dial-peer voice 6 pots
description tcatch all dial peer for BRI/PRIv
translation-profile incoming nondialable
incoming called-number .%
direct-inward-dial
dial-peer voice 50 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
direct-inward-dial
port 0/1/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
direct-inward-dial
port 0/1/1
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
translation-profile outgoing XFER_TO_VM_PROFILE
destination-pattern 98
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 2001 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 97
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 2012 voip
description ** cue prompt manager number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 96
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (VTX) **
session protocol sipv2
session target sip-server
incoming called-number .%
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1001 voip
corlist outgoing call-local
description ** star code to SIP trunk (VTX) **
destination-pattern *..
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1003 voip
description ** Passthrough Inbound Calls for PSTN from CUE **
translation-profile incoming SIP_Passthrough
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number ABCDT
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1005 voip
description ** Passthrough Inbound Calls for MWI from CUE **
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number A80T
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1009 voip
description ** Passthrough Inbound Calls for Internal Extensions from CUE **
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number ^..$
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1033 voip
corlist outgoing call-local
description **CCA*Switzerland*Short Code Services**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 0187
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1042 voip
corlist outgoing call-emergency
description **CCA*Switzerland*Ambulance / Poisioning**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 0014[45]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1041 voip
corlist outgoing call-emergency
description **CCA*Switzerland*REGA Air Rescue**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 00333333333
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1025 voip
corlist outgoing call-national
description **CCA*Switzerland*National Destination Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00[789]1.......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1020 voip
corlist outgoing call-national
description **CCA*Switzerland*Regional Announcement VM**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 01600
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1040 voip
corlist outgoing call-emergency
description **CCA*Switzerland*REGA Air Rescue**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 000333333333
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1043 voip
corlist outgoing call-emergency
description **CCA*Switzerland*Ambulance / Poisioning**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 014[45]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1035 voip
corlist outgoing call-national
description **CCA*Switzerland*Mobile Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 007[46789].......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1024 voip
corlist outgoing call-national-plus
description **CCA*Switzerland*Personal Numbering**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00878......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1029 voip
corlist outgoing call-national
description **CCA*Switzerland*Voicemail Access**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00860.........
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1036 voip
corlist outgoing call-national
description **CCA*Switzerland*VPN Access**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00869.............
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1027 voip
corlist outgoing call-national-plus
description **CCA*Switzerland*Premium Rate (Business)**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00900......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1026 voip
corlist outgoing call-national
description **CCA*Switzerland*Test Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00868T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1034 voip
corlist outgoing call-national-plus
description **CCA*Switzerland*Shared Cost numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 0084[0248]......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1038 voip
corlist outgoing call-emergency
description **CCA*Switzerland*Emergency**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 0011[278]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1037 voip
corlist outgoing call-toll-free
description **CCA*Switzerland*Toll Free Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00800......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1039 voip
corlist outgoing call-emergency
description **CCA*Switzerland*Emergency**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 011[278]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1032 voip
corlist outgoing call-national
description **CCA*Switzerland*National Destination Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00[23456]........
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1023 voip
corlist outgoing call-international
description **CCA*Switzerland*International Calls**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 000T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1031 voip
description **CCA*Switzerland*Premium Rate (Social)**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 0090[16]......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1030 voip
corlist outgoing call-national
description **CCA*Switzerland*Short Code**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 014[0357]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1045 voip
corlist outgoing call-emergency
description **CCA*Switzerland*REGA/Glaciers Air Rescue**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 0141[45]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1028 voip
corlist outgoing call-national-plus
description **CCA*Switzerland*Directory Enquiries**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 018[15].
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1021 voip
corlist outgoing call-national
description **CCA*Switzerland*Short Code**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 011[45].
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1022 voip
corlist outgoing call-national
description **CCA*Switzerland*Short Code Services**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 01[67].
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1044 voip
corlist outgoing call-emergency
description **CCA*Switzerland*REGA/Glaciers Air Rescue**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 00141[45]
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 2002 voip
description ** cue voicemail PSTN number **
translation-profile outgoing VM_Profile
destination-pattern xxx$
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 2003 voip
description ** cue auto attendant PSTN number **
translation-profile outgoing AA_Profile
destination-pattern xxx$
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1110 pots
preference 9
destination-pattern xxx
port 0/0/0
no sip-register
dial-peer voice 3006 voip
description SIP
translation-profile incoming SIP_Called_9
session protocol sipv2
session target sip-server
incoming called-number xxx.
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
no dial-peer outbound status-check pots
sip-ua
keepalive target dns:site1.365873.trk.ipvoip.ch
authentication username xxx password 7 xxx
no remote-party-id
retry invite 2
retry register 10
timers connect 100
timers keepalive active 100
registrar dns:site1.365873.trk.ipvoip.ch expires 3600
sip-server dns:site1.365873.trk.ipvoip.ch
host-registrar
telephony-service
sdspfarm units 5
sdspfarm transcode sessions 10
sdspfarm tag 2 mtpa4934c6ee4e0
video
fxo hook-flash
max-ephones 40
max-dn 300
ip source-address 10.1.1.1 port 2000
auto assign 1 to 1 type bri
calling-number initiator
service phone videoCapability 1
service phone ehookenable 1
service phone ehookEnable 1
service dnis overlay
service dnis dir-lookup
service dss
timeouts interdigit 5
system message SwissT.Net
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.1/voiceview/authentication/authenticate.do
cnf-file location flash:
cnf-file perphone
user-locale U4 load CME-locale-de_DE-German-8.1.2.2.tar
network-locale U4
load 521G-524G cp524g-8-1-17
load 525G spa525g-7-5-4
load 501G spa50x-30x-7-5-2b
load 502G spa50x-30x-7-5-2b
load 504G spa50x-30x-7-5-2b
load 508G spa50x-30x-7-5-2b
load 509G spa50x-30x-7-5-2b
load 525G2 spa525g-7-5-4
load 301 spa50x-30x-7-5-2b
load 303 spa50x-30x-7-5-2b
time-zone 23
time-format 24
date-format dd-mm-yy
keepalive 30 auxiliary 4
voicemail 98
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
hunt-group logout HLog
moh flash:/media/music-on-hold.au
multicast moh 239.10.16.16 port 2000
web admin system name cisco secret 5 xxx
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern .T
transfer-pattern 0.T
transfer-pattern 6.. blind
secondary-dialtone 0
night-service day Sun 17:00 09:00
night-service day Mon 17:00 09:00
night-service day Tue 17:00 09:00
night-service day Wed 17:00 09:00
night-service day Thu 17:00 09:00
night-service day Fri 17:00 09:00
night-service day Sat 17:00 09:00
fac standard
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-template 1
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
service phone webAccess 0
softkeys remote-in-use Newcall
softkeys idle Redial Pickup Mobility Newcall Cfwdall Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Trnsfer Mobility TrnsfVM Confrn Acct Park
button-layout 7931 2
ephone-template 15
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
softkeys remote-in-use Newcall
softkeys idle Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
button-layout 7931 2
ephone-template 16
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
softkeys remote-in-use Newcall
softkeys idle Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
ephone-template 17
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
softkeys remote-in-use CBarge Newcall
softkeys idle Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
ephone-template 18
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
softkeys remote-in-use CBarge Newcall
softkeys idle Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
button-layout 7931 2
ephone-dn 9
number BCD no-reg primary
description MoH
moh out-call ABC
ephone-dn 292
number xxx
description SIP Main Number registration
preference 10
ephone-dn 293 dual-line
number 90 secondary xxx no-reg both
label Zentrale
description 90
name Zentrale
call-forward busy 98
call-forward noan 98 timeout 20
ephone-dn 294 dual-line
number 94 secondary xxx no-reg both
label LL
description Lehrling Lehrnende
name Lehrling Lehrnende
mobility
snr xxx delay 1 timeout 30 cfwd-noan 98
snr ring-stop
call-forward busy 98
call-forward noan 98 timeout 20
ephone-dn 295 dual-line
number 93 secondary xxx no-reg both
label CM
description
name
snr xxx delay 1 timeout 30 cfwd-noan 98
snr ring-stop
call-forward busy 98
call-forward noan 98 timeout 10
ephone-dn 296 dual-line
number 92 secondary xxx no-reg both
label EE
description
name
mobility
call-forward busy 98
call-forward noan 98 timeout 20
ephone-dn 297 dual-line
number 91 secondary xxx no-reg both
label RS
description
name
mobility
snr xxx delay 1 timeout 30 cfwd-noan 98
snr ring-stop
call-forward busy 98
call-forward noan 98 timeout 10
ephone-dn 298
number 6.. no-reg primary
description ***CCA XFER TO VM EXTENSION***
call-forward all 98
ephone-dn 299
number A801.. no-reg primary
mwi off
ephone-dn 300
number A800.. no-reg primary
mwi on
ephone 1
device-security-mode none
mac-address A44C.11A0.B648
ephone-template 1
max-calls-per-button 2
username "xxx" password xxx
type 525G2
button 1:296 2:293 3m297 4m295
button 5m294
ephone 2
device-security-mode none
mac-address A44C.11A0.B566
ephone-template 1
max-calls-per-button 2
username "xxx" password xxx
type 525G2
button 1:297 2:293 3m296 4m295
button 5m294
ephone 3
device-security-mode none
mac-address A44C.11A0.B5C4
ephone-template 1
max-calls-per-button 2
username "xxx" password xxx
type 525G2
button 1:295 2:293 3m297 4m296
button 5m294
ephone 4
device-security-mode none
mac-address A44C.11A0.B67A
ephone-template 1
max-calls-per-button 2
username "xxx" password xxx
type 525G2
button 1:294 2:293 3m297 4m296
button 5m295
alias exec cca_voice_mode PBX
alias exec cca_vm_notification schedule from_time=00 to_time=24
alias exec clid-ALL_BRI ;1:0-4;1:0-9;1:0-9;1:1-9
alias exec clid-SIP ;1:1-9;1:1-9;1:1-9
banner login ^CCisco Configuration Assistant. Version: 3.2 (3). Fri Jul 04 13:18:33 CEST 2014^C
line con 0
no modem enable
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
line vty 0 4
transport preferred none
transport input all
line vty 5 100
transport preferred none
transport input all
ntp master
ntp server 91.240.0.5 prefer
enHi Patrick
I am working on this one as well. I have a UC560 with SIP Trunk provider Les.NET.
It was working fine until a few weeks ago when something changed on the provider end and broke it. My hunch it is something to do with the SIP REFER.
http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-express/91535-cme-sip-trunking-config.html
Here is an excerpt from the above page:
Call Transfer
When a call comes in on an SIP trunk to an SCCP Phone or CUE AutoAttendant (AA) and is transferred, the CME by default will send a SIP REFER message to the SP proxy. Most SP Proxy Servers do not support the REFER method. This needs to be configured in order to force the CME to hairpin the call:
Router(config)#voice service voip
Router(conf-voi-serv)#no supplementary-service sip refer
Figure 3 shows the behavior of the CME system with the REFER method disabled. -
Unable to place call on calls on hold - SIP Trunk from CUCM to CUBE and from CUBE to ISTP
Hi Cisco Community,
I have a SIP Trunk setup between the CUCM and CUBE and another SIP Dial Peers from the CUBE to the ITSP. All incoming/outgoing calls, DTMF-Relay works well except one thing which is the ability to hold the call.
On the SIP Trunk from the CUCM to the CUBE, I did not select “MTP” because when I do so, I am forced to select my preferred MTP codec which when selected G.729/G.729a, all my outgoing calls goes out using G.729r8. This codec works well for most of the calls until the ITSP replies back with G.729br8. When this condition occur, my call simply fails (this is very intermittent and only some random numbers).
That said, I have some issues with DTMF Relay when I select MTP on the SIP Trunk. DTMF Relay only works if the call is G.729r8 all the way from the CUCM to the ITSP. If the ITSP replies back with G.729br8, the call might established but will simply be “voice-only”.
The current setup is no MTP is selected and everything is working perfectly. I am happy with that until I place a call on hold, which when I do so, the call immediately terminate. Could you please help me understand why?
I have all media resources configured such as G.729r8 MTP, G.729br8 MTP, G.711u MTP, Transcoding with all codecs, etc. All MRG and MRGL are configured on all devices and SIP Trunks.
Below is an example of a call that is connected with the current setup:
Note:
IP: 10.18.81.2 (CUBE)
IP: 10.18.81.11 (CUCM SUB)
IP: 10.111.111.254 (ITSP SBC)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:29.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:10.18.81.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
Session-Expires: 1800
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
Nov 30 11:44:29.942: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:29.946: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1417347869
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 301
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6958 IN IP4 10.18.81.2
s=SIP Call
c=I
PM-HO-VG-01#N IP4 10.18.81.2
t=0 0
m=audio 22256 RTP/AVP 18 0 8 101
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Nov 30 11:44:29.950: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Nov 30 11:44:30.658: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:30.662: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.226: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf9
PM-HO-VG-01#4839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,application/xml
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC81D00
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:31.634: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72075e3a02c1
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Type: application/sdp
Content-Length: 236
v=0
o=CiscoSystemsCCM-SIP 9082578 1 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.80.40
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
PM-HO-VG-01#
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
PM-HO-VG-01#sh sip
PM-HO-VG-01#sh sip-ua call
PM-HO-VG-01#sh sip-ua calls
Total SIP call legs:2, User Agent Client:1, User Agent Server:1
SIP UAC CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 27218091323
Called Number : 0862000000
Bit Flags : 0xC04018 0x10000100 0x0
CC Call ID : 64511
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.111.111.254]:5060
Destn SIP Resp Addr:Port: [10.111.111.254]:5060
Destination Name : 10.111.111.254
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64511
Stream Type : voice+dtmf (0)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22256
Media Dest IP Addr:Port : [10.111.111.254]:20074
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 0218091323
Called Number : 0862000000
Bit Flags : 0xC0401E 0x10000100 0x80004
CC Call ID : 64510
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.18.81.11]:5060
Destn SIP Resp Addr:Port: [10.18.81.11]:5060
Destination Name : 10.18.81.11
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64510
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22350
Media Dest IP Addr:Port : [10.18.80.40]:21928
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Server(UAS) calls: 1
PM-HO-VG-01#
PM-HO-VG-01#
PM-HO-VG-01#
As you can see, the call is connected and everything is working perfectly. When I press the hold button, here is what I get:
NOTE: I have # debug ccsip messages and #debug ccsip calls (running)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:49.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 9082578 2 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:49.218: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.218: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6959 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22256 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 102 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 360
v=0
o=BroadWorks 316169737 2 IN IP4 10.111.111.254
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
a=inactive
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.282: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECA2633
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.282: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2748 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22350 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72094953dfea
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
Nov 30 11:44:49.290: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Length: 0
Nov 30 11:44:49.294: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.294: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.338: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 103 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 306
v=0
o=BroadWorks 316169737 3 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:49.342: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 2
PM-HO-VG-01#00 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2749 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:49.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720b594cd517
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 213
v=0
o=CiscoSystemsCCM-SIP 9082578 3 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.81.10
t=0 0
m=audio 4000 RTP/AVP 18
a=X-cisco-media:umoh
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=sendonly
Nov 30 11:44:49.354: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 101 BYE
Reason: Q.850;cause=86
P-RTP-Stat: PS=874,OS=17480,PR=872,OR=17440,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.354: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 104 BYE
Reason: Q.850;cause=65
P-RTP-Stat: PS=872,OS=17440,PR=952,OR=19040,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 Race Condition
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
Timestamp: 1417347889
CSeq: 104 BYE
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 200
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 101 BYE
Content-Length: 0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 86
Disconnect Cause (SIP) : 200
PM-HO-VG-01#Hi Manish,
Again, excellent feedback. Much appreciated.
I will try the commands suggested above and see if I can get DTMF to work correctly while interworking H.323 and SIP.
But my ultimate goal is to have SIP all way from the CUCM to the CUBE and from the CUBE to the ITSP.
If I enable SIP Early Offer with MTP on the CUCM going to the CUBE, all SIP Invite sent from the CUCM to the CUBE uses G.729r8 as the codec and once the call is established using G.729r8 should the ITSP reply with that codec, the call succeed and I am able to see an active MTP session using G.729 when I issue the command # show sccp connections.
One thing that I saw is that my ITSP love so much sending G.729br8 most of the times, so even if using SIP EO with MTP on the SIP Trunk to the CUBE, when I sent my INVITE out from the CUCM to the CUBE using G.729r8, specially on call center numbers such as 0800 numbers, you will see that the call established but the codec being negotiated is G.729br8 which is voice only (missing DTMF).
I will be doing some intensive test again later on this week and will send the logs.
Here is my question to both of you:
Which is the best way of having a proper SIP to SIP setup all the way that will not pose any problem?
Do I have to enable Early Offer on the SIP Profile used by the CUBE SIP Trunk or should I use the normal Standard SIP Profile? Do I need to enable MTP on my CUBE SIP Trunk or not?
From the CUBE point of view, I have a voice class codec that support G.729r8 or G.729br8 and the DTMF Relay method supported by the ITSP is RFC 2833.
I will send more logs for each scenario. I think that we are getting close to the resolution of this problem.
Thanks again for your support fellows. -
Callcentric SIP Trunk (ITSP -- 2811 CUBE -- CUCM 8.6
I have a SIP trunk from call centric that goes into my lab gear - they appear to be a good sip service due to cost but I'm having some trouble getting calls to route correctly. The call flow is Callcentric.com ITSP (SIP) --> 2811 (acting as cube) -->SIP Trunk --> CUCM 8.6. Phones are registered to CUCM.
I have the sip trunk registered and calls come in to the router (I see them in ccsip message/call debugs) The 2811 running 15.1(4)M7). Callcentric sends the username of the customer in the sip Invite instead of the called number, the called number is in the TO field. I have several DID’s from Callcentric (18452055544, 18452055545, 18452055546) for my lab. There are a few configs on here for CME where the customer number (17772253754) is simply translated to their phone DN - which is fine if you only have 1 DN with callcentric but more than 1 and thats not feasible since every inbound did will be matched to that 17772253754 translation/phone dn.
I’m using the a guide from http://tblog.cisco.be/2011/02/17/cube-conditional-sip-profiles/ using the Copy function as described http://www.cisco.com/c/en/us/products/collateral/ios-nx-os-software/ios-software-release-15-1-3-t/product_bulletin_c25-635704.html
I haven’t been able to find anything where they actually explain all the header fields so Its mostly trial and error.. so far mostly error. I think I’m close.. but who knows. Any assistance would be greatly appreciated
voice class sip-profiles 1
request INVITE peer-header sip TO copy ".sip:(.*)@." u01
request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
CUCM (single/pub)- 192.168.1.200
2811 acting as cube - 192.168.1.203
Calling Number - 18165297500
Called Number - 18452055544
vrtr1#show sip register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
17772253754 -1 20 yes
vrtr1#
The Call Setup Information is:
Call Control Block (CCB) : 0x49646C28
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 18165297500
Called Number : 17772253754 (my customer number not called number)
Source IP Address (Sig ): 192.168.1.203 (my 2811 router)
Destn SIP Req Addr:Port : 204.11.192.159:5080
Destn SIP Resp Addr:Port : 204.11.192.159:5080
Destination Name : 204.11.192.159
Feb 14 11:20:53.303: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
f: <sip:[email protected]>;tag=3601387252-874282
t: <sip:[email protected]>
i: [email protected]
CSeq: 1 INVITE
Max-Forwards: 8
m: <sip:[email protected]:5080;transport=udp>
Supported: timer
c: application/sdp
l: 350
v=0
o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.159
s=sip call
c=IN IP4 204.11.192.159
t=0 0
m=audio 61094 RTP/AVP 18 0 8 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
From: <sip:[email protected]>;tag=3601387252-874282
To: <sip:[email protected]>
Date: Fri, 14 Feb 2014 17:20:53 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
From: <sip:[email protected]>;tag=3601387252-874282
To: <sip:[email protected]>;tag=35399D8-63
Date: Fri, 14 Feb 2014 17:20:53 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=1
Content-Length: 0
Feb 14 11:20:53.419: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
f: <sip:[email protected]>;tag=3601387252-874282
t: <sip:[email protected]>;tag=35399D8-63
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 10
l: 0
u all
Feb 14 11:20:57.067: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:18452055544;cic=0288;rn=6465471001;[email protected]:5070 SIP/2.0
v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-6bceae47efe9f53b4234698a32ac8beb
f: <sip:[email protected]>;tag=3601387252-874282
t: <sip:[email protected]>;tag=35399D8-63
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 8
l: 0
************************** Running Config **************************
sh run
vrtr1#sh running-config
Building configuration...
Current configuration : 4189 bytes
! Last configuration change at 00:34:03 CST Fri Feb 14 2014
! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
version 15.1
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
no service password-encryption
hostname vrtr1
boot-start-marker
boot system flash:
boot system flash flash:c2800nm-ipvoicek9-mz.151-4.M7.bin
boot-end-marker
card type t1 0 0
logging buffered 4096 notifications
enable password cisco
no aaa new-model
memory-size iomem 5
clock timezone CST -6 0
clock summer-time CST recurring
no network-clock-participate wic 0
dot11 syslog
ip source-route
ip cef
ip name-server 192.168.1.9
no ipv6 cef
multilink bundle-name authenticated
voice service voip
ip address trusted list
ipv4 192.168.1.0 255.255.255.0
ipv4 204.11.192.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
registrar server expires max 1800 min 1800
localhost dns:callcentric.com
outbound-proxy dns:callcentric.com
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice class sip-profiles 1
request INVITE peer-header sip TO copy ".sip:(.*)@." u01
request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2811 sn FTX1133A4QR
controller T1 0/0/0
cablelength long 0db
interface FastEthernet0/0
description ** LAN **
ip address 192.168.1.203 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.1.203
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 192.168.1.1
snmp mib persist circuit
control-plane
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/1/2
voice-port 0/1/3
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 192.168.1.200
ccm-manager config
mgcp
mgcp call-agent 192.168.1.200 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface FastEthernet0/0
mgcp bind media source-interface FastEthernet0/0
mgcp profile default
dial-peer voice 999100 pots
service mgcpapp
port 0/1/0
dial-peer voice 999101 pots
service mgcpapp
port 0/1/1
dial-peer voice 999102 pots
service mgcpapp
port 0/1/2
dial-peer voice 999103 pots
service mgcpapp
port 0/1/3
dial-peer voice 999010 pots
service mgcpapp
port 0/1/0
dial-peer voice 6 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 17772253754
voice-class sip profiles 1
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 7 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 1845205554[4-5]
voice-class sip profiles 1
dtmf-relay h245-alphanumeric
no vad
sip-ua
credentials username 17772253754 password 7 106C1B49111F17194D realm callcentric.com
authentication username 17772253754 password 7 08035E1E1D11000553 realm callcentric.com
no remote-party-id
retry invite 2
retry register 10
timers connect 100
mwi-server dns:callcentric.com expires 3600 port 5060 transport udp
registrar dns:callcentric.com expires 3600
sip-server dns:callcentric.com
host-registrar
line con 0
line aux 0
line vty 0 4
password cisco
login
transport input all
scheduler allocate 20000 1000
ntp server 199.102.46.72
ntp server 23.227.162.123 prefer
end
exitThank you for the reply. I've updated the dial-peers as sugested. I'm now seeing an invite go out to my CUCM however the call fails with a 403 (forbidden) which appears to come from the ITSP (Callcentric). I've included a new set of ccsip message debugs and the dial-peers as adjusted. Please let me know what you think.
dial-peer voice 6 voip
description ## INBOUND CALL from ITSP ##
session protocol sipv2
session target sip-server
incoming called-number 17772253754
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
dial-peer voice 100 voip
description ## INBOUND DID to CUCM ##
destination-pattern 17772253754
session protocol sipv2
session target ipv4:192.168.1.200
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
dial-peer voice 7 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 1845205554[4-5]
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
Feb 15 10:18:11.424: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
f: ;tag=3601469891-655
t: [email protected]>
i: [email protected]
CSeq: 1 INVITE
Max-Forwards: 8
m:
Supported: timer
c: application/sdp
l: 350
v=0
o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.164
s=sip call
c=IN IP4 204.11.192.164
t=0 0
m=audio 61782 RTP/AVP 18 0 8 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
Feb 15 10:18:11.456: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
From: ;tag=3601469891-655
To: [email protected]>
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Feb 15 10:18:11.460: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392481091
Contact:
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 7
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273
v=0
o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
s=SIP Call
c=IN IP4 192.168.1.203
t=0 0
m=audio 18168 RTP/AVP 18 101
c=IN IP4 192.168.1.203
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Feb 15 10:18:11.552: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35;rport=57100;received=24.123.98.94
f: [email protected]>;tag=8408644-12C8
t:
i: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="8ae6b7b1cea74cf401e8a26fd3c7371b", opaque="", stale=TRUE, algorithm=MD5
l: 0
Feb 15 10:18:11.560: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Feb 15 10:18:11.560: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1392481091
Contact:
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="17772253754",realm="callcentric.com",uri="sip:[email protected]:5060",response="a381f10fbbfbd255b444569fef0dddfe",nonce="8ae6b7b1cea74cf401e8a26fd3c7371b",opaque="",algorithm=MD5
Max-Forwards: 7
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273
v=0
o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
s=SIP Call
c=IN IP4 192.168.1.203
t=0 0
m=audio 18168 RTP/AVP 18 101
c=IN IP4 192.168.1.203
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Feb 15 10:18:11.648: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Incorrect Authentication
v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3;rport=57100;received=24.123.98.94
f: [email protected]>;tag=8408644-12C8
t:
i: [email protected]
CSeq: 102 INVITE
l: 0
Feb 15 10:18:11.660: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Feb 15 10:18:11.660: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
From: ;tag=3601469891-655
To: [email protected]>;tag=8408714-B60
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=57
Content-Length: 0
Feb 15 10:18:11.752: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
apsc-vrtr1#ACK sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
f: ;tag=3601469891-655
t: [email protected]>;tag=8408714-B60
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 10
l: 0
vrtr1#u al
Feb 15 10:18:14.776: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:18452055544;cic=0288;rn=6465471001;[email protected]:5070 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-e437c2c5cac5f1a6e147c1cd7c98aad7
f: ;tag=3601469891-655
t: [email protected]>;tag=8408714-B60
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 8
l: 0 -
DTMF tones from CUCUM 9 thru H323 GW out SIP trunk not working
This is the setup. Currently in lab environment for a client, but needs to go into production
IP Phone -> CUCM 9 -> H323 GW -> SIP Trunk -> Proprietary device -> Analog phone
Calls complete both ways with no issues. Proprietary devices only uses G711ulaw, so I have configured a xcoder on the H323 GW to transcode to G729 across the WAN link (between the CUCM cluster and the H323 GW).
Pressing keys/sending DTMF tones from the IP phone are not heard in the analog phone
Running a debug voice ccpai inout at the H323 gateway shows me that the DTMF tones are being received the GW and are being sent along. See below:
Seaport#
Seaport#
Seaport#! Pressing digit "9" on VoIP phone
Seaport#
Seaport#
Seaport#
Seaport#
*Nov 5 15:41:57.637: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 9 to dstCallId 0x49E
*Nov 5 15:41:57.637: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
*Nov 5 15:41:57.713: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 9 to dstCallId 0x49E
*Nov 5 15:41:57.713: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
Seaport#
Seaport#! Pressing digit "9" on VoIP phone " on VoIP phone 5" on VoIP phone
Seaport#
Seaport#
Seaport#
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
Seaport#
Seaport#! Pressing digit " 5" on VoIP phone
Seaport#
Seaport#
Seaport#
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
Seaport#
Seaport#
However, debug ccsip does not give me any indications that the DTMF tone is being sent out the SIP trunk. Debug ccsip all attached.
Relevant portions of the H323 configuration are below
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g729br8
interface Loopback0
ip address 172.16.88.254 255.255.255.255
ip pim sparse-dense-mode
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.88.254
interface GigabitEthernet0/1
ip address 192.168.200.254 255.255.255.0
duplex auto
speed auto
interface Loopback0
ip address 172.16.88.254 255.255.255.255
ip pim sparse-dense-mode
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.88.254
interface GigabitEthernet0/1 <- interface to proprietary device
ip address 192.168.200.254 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/2 <-interface to Local LAN supporting IP Phones
ip address 10.10.10.254 255.255.255.0
duplex auto
speed auto
sccp local GigabitEthernet0/2
sccp ccm 10.10.10.254 identifier 1 priority 1 version 3.1
sccp ccm group 1
bind interface GigabitEthernet0/2
associate ccm 1 priority 1
associate profile 10 register xcoder_1
dspfarm profile 10 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 10
associate application SCCP
dial-peer voice 2 voip
description Default Incoming Dial Peer
incoming called-number .
voice-class codec 1
dtmf-relay h245-alphanumeric h245-signal rtp-nte
dial-peer voice 6 voip
destination-pattern 90052.. <- DN of analog phone
session protocol sipv2
session target ipv4:192.168.200.1 <- IP of proprietary device
codec g711ulaw
no vad
sip-ua
registrar ipv4:172.16.88.254 expires 3600
no transport tcp
telephony-service
sdspfarm units 4
sdspfarm transcode sessions 2
sdspfarm tag 1 xcoder_1
I also ran the debug voip rtp session named-event all but nothing was displayed when I pressed the digits on the IP Phone.
JeffPlease configure "dtmf-relay rtp-nte" command under SIP dial-peers.
Jorge Armijo
Please remember to rate helpful responses and identify helpful or correct answers. -
Cucm , Cube via Sip and Sip Trunk to ISP , Outgoing calls not working
Hi
We have issue with the outgoing calls to sip trunk
Below is the config and the debugs
It will be great if you give your thoughts since we have stuck here
My thoughts are:
i see that for unknown reason the called number is going with 4 digits instead of 8 digits
i dont see any sip message comming from ISP
Maybe the call not going there ? to isp trunk? From the trace the call hit the correct dialpeer 888 but i see 4 digits as a called number , but i dodnt understant the reason to translated in 4 digits the called number.Not apply a translation rule for that
confused!!!
Calling Numbner:22324086
Called Number: 23823690
CUCM:192.168.1.241 and 242
CUBE:192.168.1.10
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*\(....\)/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^\(9\)\(.......$\)/ /99\2/
rule 3 /^\(2\)\(.......$\)/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*\(....\)/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^\(9\)\(.......$\)/ /99\2/
rule 3 /^\(2\)\(.......$\)/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vadHi Aok
I change the default value for IPVMS from g711ulaw to g711alaw but the results remained the same
Also i have restarted the IPVMS
SIP-GW#
SIP-GW#
*Mar 5 14:19:57.854: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 38874 2 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:64000
b=AS:64
t=0 0
m=audio 24784 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:57.878: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1362493197
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3807 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 101
c=IN IP4 10.249.13.130
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.878: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:57.926: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 102 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 213
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 2 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=inactive
*Mar 5 14:19:57.942: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5774 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 101
c=IN IP4 192.168.1.10
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.946: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3562A4
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:57.946: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK798246ab3597
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
*Mar 5 14:19:58.146: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Length: 0
*Mar 5 14:19:58.158: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1362493198
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:58.158: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:58.218: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 103 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 216
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 3 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 18 96 99
a=rtpmap:96 AMR/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
a=sendrecv
*Mar 5 14:19:58.234: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 283
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5775 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 18 101
c=IN IP4 192.168.1.10
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:58.242: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7985648033f2
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 192
v=0
o=CiscoSystemsCCM-SIP 38874 3 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 192.168.1.241
t=0 0
m=audio 4000 RTP/AVP 8
a=X-cisco-media:umoh
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendonly
*Mar 5 14:19:58.262: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK358582
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3808 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 99
c=IN IP4 10.249.13.130
a=sendonly
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
SIP-GW#
SIP-GW#sh voip rtp connections
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 716 717 19314 4000 192.168.1.10 192.168.1.241
2 717 716 19234 54932 10.249.13.130 10.224.42.72
Found 2 active RTP connections -
Lync 2013 with SIP trunk with panasonic kx-tde200
Hi
My company has installed a panasonic ip-pbx kx-tde for multiline with 100 number range for telephone service.
Now my company is going to replace multiline by sip trunk . It will still work with Panasonic pbx box just need to reprogramme to be able to connected to the sip proxy which is managed by internet service provider.
For this scenario , would Lync 2013 voice work if I just add PSTN gateway which is the ip of panasonic pbx address to the frontend in topology ? Or I may need a mediation server as a must requirement to make lync voice work?
Thanks
WenFeiMedia bypass allow a call to basically skip the mediation server once it's established and go directly from gateway (in this case the PBX) and the endpoint (the telephone handset or Lync client) More information here: http://technet.microsoft.com/en-us/library/gg398719.aspx
By having this (if your PBX supports it) you reduce the load on the mediation servers. Before you go too far down the road also make sure that your PBX supports SIP trunks that are SIP over TCP (as Lync doesn't work with SIP over UDP)
Sort of, the easiest way is to add the .com as an additional SIP domain in Topology builder, you will need to create DNS records for it (both internal and external) and you will need to reissue the certs with additional SANs to support the second domain.
YOu will also need to update all the users to use the new suffix of xxx.com. So it's not a small task.
If this helped you please click "Vote As Helpful" if it answered your question please click "Mark As Answer" | Blog
www.lynced.com.au | Twitter
@imlynced -
SIP phone registering on SIP trunk
Hi,
i have a UC 500 connected to our phone provider using a SIP trunk.
All the phones are SPA508 G
All is working fine !
Then, some days ago i added a SIP phone (extention 350) on the UC500, that also worked fine, and then after some minutes all our incoming/outgoing calls were blocked.
I called my provider that told me that our IP was banned because they have seen to much registration attempt from a bad user that was "350"
I can confirm with a "sh sip-ua register status" command that i had two sip registration : my SIP trunk and the SIP phone
Then it seems that the UC 500 is trying to register the SIP phone on the SIP trunk ?
What am i doing wrong ?
Is there a command to avoid that ?
Bellow is how the SIP phone and the SIP trunk are configured
Many thanks for your help, i was unable to find anything about that, but i guess somebody already had this problem !
The SIP phone -------------------------------------------------------------------------
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
fax protocol none
modem passthrough nse codec g711ulaw
sip
registrar server expires max 3600 min 120
no update-callerid
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
voice register global
mode cme
source-address 10.1.1.1 port 5060
max-dn 20
max-pool 20
load 9971 sip9971.9-2-2
load 9951 sip9951.9-2-2
load 8961 sip8961.9-2-2
load 7971 term71.default
authenticate register
authenticate realm xxxxxx.com
timezone 13
hold-alert
mwi stutter
mwi reg-e164
create profile sync 0636240803635305
voice register dn 1
number 350
name Conference
label Conference
voice register pool 1
id mac 1234.1234.1234
number 1 dn 1
username 350 password 1234
codec g711ulaw
The SIP trunk ----------------------------------------------------------------------
sip-ua
credentials username user1234 password 1234 realm sipgw9.provider.com
authentication username user1234 password 1234 no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:sipgw9.provider.com expires 3600
sip-server dns:sipgw9.provider.comI'm still searching on the forum, and maybe i found somthing related to my problem, not sure... any advice ?
Disable outbound proxy on voice register global as by default it will use the outbound proxy configured on the system which would not make sense
voice register global
no outbound-proxy
found there : https://supportforums.cisco.com/discussion/10760741/uc500-sip-server-and-sip-trunk -
Confused by basic SIP Trunk configuration.
I've went through a few basic SIP trunk configurations and Youtube videos the last couple days but can't figure out what I'm doing wrong.
I've set up H323 and MGCP no problem, but I can't figure out the SIP trunk set up. I'm guessing there are some concepts I'm not understanding yet.
I've got a CUCM lab set up. A 2851 PSTN Simulator, 2851 H323 Gateway at the Main site with a 9.0 CUCM setup in that site and a Branch site that I'm trying to set up as a SIP trunk to connect two phones.
CUCM is on the 192.168.5.x/24 subnet. 172.16.0.x/24 is the subnet connecting the serial(internet) cable between the two gateways in which I'm trying to establish the trunk between.
The Branch phones are still registering with the CUCM at the main site. The Route Pattern is looking to the Branch Route List which has the SIP Trunk listed. I'm just getting a fast busy when trying to place a call from the branch site to the main site.
The most frustrating thing I'm not understanding, is that the debug ccsip and call debugs on my SIP Branch gateway shows absolutely nothing. I've tried registering the branch phones with the SIP Trunk, but stopped when I figured that shouldn't be necessary.
If someone can make some sense of this, I'd truly appreciate it!Hello Aditya and thanks for the consideration!
I do have a direct IP connection, but I want to set up a SIP trunk and use it just to know how to do it before I do it in production.
I did end up deleting the phones from CUCM so they can register with the 2851 CME that I'm setting up as a SIP trunk. So it is registering there, and I set the allow connections and bind sip commands.
I am now getting Debugs and calls from the SIP Trunk router going to CUCM, but the error message is No Codec, and I Get the fast busy after the call rings on the CUCM Main Site side. So looks like the negotiation is failing. Here is my CLI for the SIP Trunk now after the changes have been made and phones registered to the SIP Branch site as well as the Debug when I tried to place a call to extension "5000":
Note: I did try to change the codecs in the dial-peers to g729r8 instead of 711 and same fast busy after answering.
==============================================
Branch_SIP#show run
Building configuration...
Current configuration : 3529 bytes
! Last configuration change at 03:15:11 UTC Thu Apr 2 2015
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Branch_SIP
boot-start-marker
boot-end-marker
! card type command needed for slot/vwic-slot 0/2
enable secret 5 $1$hOXF$gvfmWW1ZIQE0mAMVg.u1c/
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 10.0.10.1 10.0.10.10
ip dhcp excluded-address 10.0.30.1 10.0.30.10
ip dhcp pool Data
network 10.0.10.0 255.255.255.0
default-router 10.0.10.254
option 150 ip 192.168.5.250
dns-server 192.168.5.200
ip dhcp pool Voice
network 10.0.30.0 255.255.255.0
default-router 10.0.30.254
dns-server 192.168.5.200
option 150 ip 172.16.0.1
ip dhcp pool data
option 150 ip 172.16.0.2
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
bind media source-interface Loopback1
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2851 sn FTX1031A2FM
redundancy
interface Loopback1
ip address 2.2.2.2 255.255.255.255
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.10
encapsulation dot1Q 10
ip address 10.0.10.254 255.255.255.0
interface GigabitEthernet0/0.30
encapsulation dot1Q 30
ip address 10.0.30.254 255.255.255.0
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Serial0/3/0
no ip address
shutdown
clock rate 2000000
interface Serial0/3/1
ip address 172.16.0.1 255.255.255.0
clock rate 250000
interface Internal-Service-Module0/0
no ip address
shutdown
!Application: CUE Running on AIM2
hold-queue 512 out
router eigrp 1
network 0.0.0.0
network 2.2.2.2 0.0.0.0
network 10.0.0.0
network 10.0.10.0 0.0.0.255
network 10.0.30.0 0.0.0.255
network 172.16.0.0
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 172.16.0.2
tftp-server flash:term45.default.loads
tftp-server flash:jar45sccp.8-5-3TH1-6.sbn
tftp-server flash:cnu45.8-5-3TH1-6.sbn
tftp-server flash:apps45.8-5-3TH1-6.sbn
tftp-server flash:dsp45.8-5-3TH1-6.sbn
tftp-server flash:cvm45sccp.8-5-3TH1-6.sbn
control-plane
voice-port 0/0/0
voice-port 0/0/1
mgcp profile default
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
session protocol sipv2
session target sip-server
codec g711ulaw
dial-peer voice 2 voip
description **Outgoing Call to SIP Trunk**
destination-pattern 5...
session protocol sipv2
session target sip-server
codec g711ulaw
sip-ua
sip-server ipv4:192.168.5.250
telephony-service
codec g711ulaw
max-ephones 24
max-dn 48
ip source-address 172.16.0.1 port 2000
system message SIP Branch Site
cnf-file location flash:
load 7960-7940 P00308010200.bin
max-conferences 8 gain -6
transfer-system full-consult
ephone-dn 1
number 4008
ephone-dn 2
number 4005
ephone 1
device-security-mode none
mac-address 001D.A21A.2065
button 1:1
line con 0
exec-timeout 0 0
line aux 0
line 194
no activation-character
no exec
transport preferred none
transport input all
transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
speed 115200
line vty 0 4
password cisco
login
transport input all
line vty 5 15
password cisco
login
transport input all
scheduler allocate 20000 1000
end
Branch_SIP#show debug
TFTP:
TFTP Event debugging is on
CCSIP SPI: SIP Call Statistics tracing is enabled (filter is OFF)
Branch_SIP#
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4B6C5C28
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 4008
Called Number : 5005
Source IP Address (Sig ): 172.16.0.1
Destn SIP Req Addr:Port : 192.168.5.250:5060
Destn SIP Resp Addr:Port : 192.168.5.250:5060
Destination Name : 192.168.5.250
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 2.2.2.2
Source IP Port (Media): 19472
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 63
Disconnect Cause (SIP) : 503
Branch_SIP#
Maybe you are looking for
-
Apple's App Store + Flash + PC Only --Info here! ;)
*I do not own a MAC at all and used Flash CS5 + Air 2.6 SDK to make APPs for the iOS and then upload them to the APP Store to try and make money!* Hey guys, I just want to say first that if you have a PC and you have Flash CS5, and you have a deve
-
First DTP in Process chain giving Error "CX_RSB_WRITE_ERROR"
Hi All, I am facing a problem when I execute a process chain. I have a master process chain and 3 sub process chains. Each of these sub process chains are having some DTP in it in a sequential order. These DTP's are Independent of each other, so even
-
Making PAN Number Field MANDATORY in Vendor Master while CREATING VENDOR
Dear Experts, I want to create a Vendor Master with PAN Number Field Mandatory. Is there any input for making PAN Number Field Mandatory at the time of creating a vendor. Regards - Manjunath
-
2 Decimal points in inventory module
Dear Expert, In SAP 2005, inventory module allowed for 6 decimal points somehow in SAP 2007, it does not allow for 6 decimal points only 2 decimal points are allowed. When I try to zerolise my Inventory quantity, they still remain some quantity in t
-
Fuzzy Lookup Add-In for Excel - Unattended installation
Hi. I would lige the Fuzzy Lookup Add-In for Excel to be installed silently, but it seems only to be possible for a Per-User installation with the parameter "setup.exe /passive" I need the installation to be for All Users, but I can´t find any parame