Inbound SIP trunk busy when routed to AA
Hey guys,
We have been having some strange things with our UC520 lately, so I built up a UC540 as a backup, and then rebuilt our UC520. Both of these systems are exhibiting the same behavior on all inbound SIP calls that are routed directly to our AA - a fast busy signal.
At the end of the day, I want all incoming SIP calls to go to a blast group during the day, and our After Hours AA . . . well, after hours. The way that I would like to accomplish this is through a combination of Floating Extensions and Night Service, but it doesn't quite work. Ideally, my floating extension would forward all calls to my initial blast group during the day. Night Service would forward those same calls to our AA after hours. Floating extensions works to automatically forward inbound calls to my blast group, but the Night Service rules don't bypass that blast group to send the inbound calls directly to our AA. You have to wait until the cfna timer completes before you finally get the AA.
So in the past, I've just set up a CIPC where I do the same thing. Configure the CIPC to forward all calls to the blast group, then set up night service. This actually works, but right now, those inbound calls give me a fast busy whenever they are delivered to the AA extension. All other extensions seem to work okay.
It seems like this is probably a transcoding issue, but I've not been able to find it. Preferred codec on the UC is G.711ulaw.
SIP Trunk provider is NexVortex.
Any suggestions on where to start looking? Thanks for your help!
Seth
Figured it out. I had turned off "hairpinning" for the SIP trunk. It would appear that the system was therefore not forwarding the calls to the AA.
Similar Messages
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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! -
Third Party Phone over SIP Trunk with CUCM 9.x
Hi all,
I have a problem where my Third Party SIP phones wont go over the SIP trunk configured in my CUCM 9.x cluster. My Cisco phones work fine and goes out the trunk. I have noticed a distinct difference in wireshark with the invite packets from Third Party SIP phones and the Cisco ones.
I have configured the SIP trunk in CUCM with the following route pattern (60.!#)and configured it with associated group and list. Heres the differense between the invite packets from Cisco and Third Party phones.
Cisco Phone: INVITE sip.60xxxx%23@ipadress
Third Party SIP Phone: INVITE sip:[email protected]
It seems the Cisco phones gets some extra configured the Third Party ones dont...
Thanks in advance for any help.
//PerThanks for the answer
Yeah i have DNS configured and i have the trunk pointed to a domain destination SRV record and like i said it works fine when calling from a Cisco phone. I tried changing the domain to an ip address but same result. I also changed the Plycom phone from being registered towards the domain of CUCM to an IP adress of CUCM and then the SIP INVITE messages in wireshark began to look kinda the same expet for the "%23" section but it still dont work.
When i look at the Real Time Data in RTMT the orig and final called from the cisco phone has stripped the 60 and forwared the rest of the number towards the correct domain for the SIP trunk.
When looking at the data from the Polycom phone the orig and final called data still contains the 60 prefix part and the called device name field is empty. The termination Cause Code is that the number requested is Unallocated/Unassigned..
In other words something is missing to get CUCM to strip 60 from the Polycom phones dialed number and send it towards the SIP trunk like it does when the Cisco phones call it.
Unfortunatley i dont have the meens to attach the trace...
Thanks again for any help/advice
With regards, Per. -
Has anyone set up Lync server 2010 to use the Gamma SIP trunks, that dont require the use of a gateway?
No requirement for an additional gateway device, with direct MS Lync connectivity
The trouble is i cannot get Lync to connect to the trunks. We have purchased the SIP trunks from a gamma supplier(we didnt now they were a supplier, until recently when we asked for support and they went 'duhhhhhh me no know, we just
sell things dunow how to set things up' what a PAIN IN THE A***), and they say that the SIP trunks are pointed at our EFM IP address. which also has the DDIs assigned to it.
So, i setup a PSTN gateway on lync topology using IP of EFM, Listening port 5060 using TCP. Are these ports and protocol okay?
The VoIP phones seem to want to call, they just lack any sound, no ringing tone, no dissconnected tone. Just says calling "+44157322****" So the dial plan is changing 22**** to the correct local code and whatever the +44 thing
is.
Any advice on how i can find the problem, or how to setup the trunks up would be hugely appreciated.
P.S We initially tried to use an audiocodes mediant 1000, which was what we asked our trunk supplier about, and then they informed us about being a gamma supplier, and that the gamma trunks do not require a gateway. Followed setting
up guide for mediant 1000 with gamma trunks through audiocodes blah, to no success. I think thats because it was changing the coders, which was not needed if the trunks are directly compatable.Hi,
Please review the SIP trunk topology.
http://technet.microsoft.com/en-us/library/gg398720.aspx
To
implement SIP trunking, you must route the connection through a Mediation Server, which proxies communications sessions between Lync Server 2010 clients and the service provider and transcodes media when necessary. Each Mediation Server has
an internal and an external network interface. The internal interface connects to the Front End Servers. The external interface is commonly called the gateway interface because it has traditionally been used to connect the Mediation Server to a PSTN gateway
or an IP-PBX. To implement a SIP trunk, you connect the external interface of the Mediation Server to the external edge component of the ITSP. The external edge component of the ITSP could be a Session Border Controller (SBC), a router, or a gateway.
Generally the gateway is not required in your organization. You need to configure Mediation Server setting. For the details about
the SIP trunk configuration of ITSP side, you need to contact Gamme Support for further assistance.
Regards,
Kent Huang
TechNet Community Support ************************************************************************************************************************ Please remember to click “Mark as Answer” on the post that helps you, and to click “Unmark as Answer” if a
marked post does not actually answer your question. -
Route SIP REFER to SIP Trunk based on DN
Cisco UCM 9 is connected to a third-party PBX over SIP Trunk. Third-party PBX sends a SIP REFER message to Cisco UCM to call a DN on the third-party PBX. Cisco UCM responds with SIP 404 Not Found as it does not recognize the DN of the third-party PBX.
How do I configure Cisco Unified Communication Manager 9 to route this call back out over the SIP Trunk to the third-party PBX based on the DN (Not IP)?
Cisco UCM contains a route pattern 53xxx to route to SIP_Trunk_3rdParty.
Third-party PBX contains a SIP Proxy and Call Server. The call should route to the SIP Proxy IP. The SIP REFER contains "Refer-To" 53xxx@ThirdPartyCallServerIP
I added a SIP Route Pattern on CUCM to route calls for ThirdPartyCallServerIP to SIP_Trunk_3rdParty. This works in routing the call to ThirdPartyCallServerIP, however I need the call to route to 53xxx@ThirdPartySIPproxyIP for it to be successful.
Direct calls from CUCM to ThirdParty PBX 53XXX@ThirdPartySIPproxyIP are successful. SIP REFER coming into CUCM to request CUCM to call ThirdParty fail.
Any ideas on what configuration on CUCM I could try to get CUCM to route the call to thrid-party based on the SIP REFER?Thanks for the reply Vivek.
Partitions:
- ThirdPartyPBX
- CiscoEndpoints
Calling Search Space: "ThirdParty_Cisco" contain both of the above partitions.
Route Pattern 531XX and 80965 are assigned to Route Partition "ThirdPartyPBX"
Cisco UCM Main site phones are in CSS "ThirdParty_Cisco" and DN is in Route Partition "CiscoEndpoints". DN is in CSS "ThirdParty_Cisco".
Trunk "SIP_Trunk_3rdParty" - Inbound and Outbound Calls are in CSS "ThirdParty_Cisco".
Trunk SIP information has "Rerouting CSS", "Out-of-Dialog Refer CSS", and Subscribe CSS as "ThirdParty_Cisco".
Cisco continues to respond to with SIP 404 not found. CUCM does not seem to match the SIP refer to the CSS or Route partition with with 531XX route pattern.
The SIP Refer is coming from DN 80965 over the SIP Trunk from the Third-party PBX.
Perhaps I'm missing something in my CSS config?
Any other method for CUCM to match SIP Refer to a Route Pattern? -
Route pattern to SIP trunk problem
Hello, I have a 2801 router that has been configured with CME and a working SIP connection to my local ISP.
Tested with calls via CME so I know for sure that the SIP config and dial plan is fine on this gateway.
Next I wanted to try out CUCM so I set up a CUCM 8.6 box that is connected to the 2801 router to use as it's SIP gateway.
The only change I made to the gateway router config was to alter the "ip option 150" address so that the phones go to CUCM for their configs etc (which they do with no problems).
Then I set up a SIP trunk in CUCM along with a route pattern which is to use the SIP trunk within the Gateway/Route list option.
But when I make a call that matches this route pattern all I get is the intermittent beep message from the phone. I cannot route calls succesfully through it.
I have checked network connectivity and all is fine. The IP address I specfied in CUCM for the SIP trunk is simply one of the interfaces on the 2801 router and it is definitley reachable.
I also activated "debug ccsip all" on the 2801 gateway router but nothing appears. So it seems like the calls are not even reaching the 2801 gateway ?
Is the problem possibly a conflit between CME on the gateway router and my CUCM ?
Do I need to disable CME somehow on the gateway first ? Or am I not doing something correct in the CUCM config ?
Thank you kindly for any suggestions.
ps. I have attached a couple of screenshots of my config.Hello, thanks for helping.
I activated "debug voice ccapi inout" as well as "debug ccsip all" on the gateway but nothing showed up.
Therefore I deduce the call is not even making it to across the SIP trunk into the gateway router ?
As I am a newbie trying this out for the first time, it is guranteed to be something really simple.
I have included my running config from the gateway router below..
One addition I made was to add an incoming dial peer. That is "dial peer 5, description CUCM SIP trunk".
I set it up with a destination patter 2... to match my phone config on CUCM which have numbering in the 2000 range.
Sorry, I got RTMT up and running but could not get any meaningful results from it. I need to learn up on that.
I did however run a 'dialed number analysis' from CUCM direct and have attached the result. It seems the dialled number "99" is matching the route pattern OK.
So why is it not then moving down the SIP trunk to my gateway and getting picked up by the incoming dial peer ?
Thanks if you guys can offer any more help.
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot system flash:c2801-ipvoicek9-mz.151-2.T0a.bin
boot-end-marker
no aaa new-model
clock timezone nzst 13 0
dot11 syslog
ip source-route
ip dhcp pool DATA_SCOPE
network 192.168.200.0 255.255.255.0
default-router 192.168.200.1
dns-server 8.8.8.8
ip dhcp pool VOICE_SCOPE
network 192.168.100.0 255.255.255.0
default-router 192.168.100.1
option 150 ip 192.168.2.115
ip dhcp pool MGMT_SCOPE
network 192.168.1.0 255.255.255.0
default-router 192.168.1.99
ip cef
ip name-server 4.2.2.2
no ipv6 cef
multilink bundle-name authenticated
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729r8
codec preference 3 g711ulaw
codec preference 4 ilbc
voice translation-rule 1
rule 1 /^9/ //
voice translation-profile Strip9ToGetOut
translate called 1
voice-card 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-2995340181
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-2995340181
revocation-check none
crypto pki certificate chain TP-self-signed-2995340181
certificate self-signed 01
3082023E 308201A7 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 32393935 33343031 3831301E 170D3733 30363034 31393534
32305A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 39393533
34303138 3130819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100C34D C8ECBB53 E01373A3 2E286B78 2D23042B 1C8588B1 A7861899 BA1C6860
AE1D7868 2A59E3BC 54D0A457 8FFDE27F C09104E5 C7A429F3 74CD9DA8 4A980366
675CC27C CDB94838 821CC05F 2C0AC2BC D882C132 6CAA1FA6 6DA740E4 562428B1
12B741F1 A50C9246 4CC35EDA DEE1D038 3883BB35 A91ABF8B 483E4160 F5FA4B5A
9A570203 010001A3 66306430 0F060355 1D130101 FF040530 030101FF 30110603
551D1104 0A300882 06526F75 74657230 1F060355 1D230418 30168014 72119640
F3396E1F E4168086 D31D8619 0D8337FF 301D0603 551D0E04 16041472 119640F3
396E1FE4 168086D3 1D86190D 8337FF30 0D06092A 864886F7 0D010104 05000381
81003B5A 29DE3A1E C5AB6092 E8D90650 C80752FC 0AAC93FD C5DE3D69 071B08FA
D4013232 81CA07E7 15F90190 6A3AD6A0 1D05F0F2 13479568 888332A5 F81E2681
7DA44095 4D11CFB7 CA79579A 8D95DE54 7B00173C E2C50573 A310C8C9 1487FEFC
CE35B66E 9EF94CFA 8D6D6DCD ADC78132 2709F198 6DF2F0FA D80CC088 D0C4C7D1 080B
quit
license udi pid CISCO2801 sn FTX0947W07M
username xxx privilege 15 password 0 xxx
interface FastEthernet0/0
ip address 192.168.3.50 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
no ip address
duplex auto
speed auto
interface FastEthernet0/1.2
encapsulation dot1Q 2
ip address 192.168.2.1 255.255.255.0
interface FastEthernet0/1.99
encapsulation dot1Q 99
ip address 192.168.1.99 255.255.255.0
interface FastEthernet0/1.100
description voice_VLAN
encapsulation dot1Q 100
ip address 192.168.100.1 255.255.255.0
interface FastEthernet0/1.200
description data_VLAN
encapsulation dot1Q 200
ip address 192.168.200.1 255.255.255.0
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip route 0.0.0.0 0.0.0.0 192.168.3.1
logging esm config
tftp-server flash:/phone/7940-7960/P00307020200.bin alias P00307020200.bin
tftp-server flash:/phone/7940-7960/P00307020200.loads alias P00307020200.loads
tftp-server flash:/phone/7940-7960/P00307020200.sb2 alias P00307020200.sb2
tftp-server flash:/phone/7940-7960/P00307020200.sbn alias P00307020200.sbn
control-plane
mgcp fax t38 ecm
dial-peer voice 1 voip
description local_7_Digit_Calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 9[2-9]......
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 2 voip
description international_calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 900T
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 3 voip
description national_calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 90[34679].......
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 4 voip
translation-profile outgoing Strip9ToGetOut
destination-pattern 90[34679].......
dial-peer voice 5 voip
description CUCM SIP trunk
destination-pattern 2...
session protocol sipv2
session target ipv4:192.168.2.115
voice-class codec 1
sip-ua
authentication username xxxxxxxxxx password xxxxxxxx
060
telephony-service
max-ephones 10
max-dn 20
ip source-address 192.168.1.99 port 2000
load 7960-7940 P00307020200
max-conferences 4 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1 dual-line
number 1000
name Lydia Francis
ephone-dn 2 dual-line
number 1001
name Leah Francis
ephone-dn 3 dual-line
number 1002
n
ephone-dn 4 dual-line
number 1003
ephone 1
mac-address C80A.A970.01DE
type CIPC
button 2:2
ephone 2
mac-address 000C.3070.8705
button 1:1 2:15
ephone 3
mac-address 000C.8546.5954
button 1:3 2:15
line con 0
logging synchronous
line aux 0
line vty 0 4
privilege level 15
login local
transport input telnet ssh
scheduler allocate 20000 1000
ntp server 195.43.74.123
end -
SIP Trunk not accepting inbound calls
I have a CME setup using Engin as a SIP provider
I am able to dial out with no issue, however my inbound calls do not work, they divert to the Engin voicemail
My SIP registration is OK and the number is configured as the primary DN on one of my phones
Router#sh sip-ua register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
038682XXXX -1 1124 yes
101 20001 45 no
102 20003 18 no
103 20005 45 no
104 20006 45 no
I do see the call come in if I debug the dial peer, but it only seems to match an outgoing dp
I am seeing a couple of disconnect cause codes that I cant seem to find any relavent information on in the CCSIP debugs
Router#
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:4F947560
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:1800, SE Refresher:uas, Min-SE Value:1800, flags:2001
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:4F947DF8
Sep 8 18:15:15.025: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:403, container:4F947B38
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4C39C570
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0417XXXXXX
Called Number : 038682XXXX
Source IP Address (Sig ): 211.30.48.136
Destn SIP Req Addr:Port : 203.161.164.69:5060
Destn SIP Resp Addr:Port : 203.161.164.69:5060
Destination Name : 203.161.164.69
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 211.30.48.136
Source IP Port (Media): 17768
Destn IP Address (Media): 203.161.164.69
Destn IP Port (Media): 18314
Orig Destn IP Address:Port (Media): [ - ]:0
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 21
Disconnect Cause (SIP) : 403
Any Ideas
DougHi Tapan,
Firstly the topology is as follows
ISP/VOIP provider - Internet - Cable modem - 2800 CME router - IP Phone
The VM is provided by the ISP
debug ccsip messages
Sep 9 10:21:41.488: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Contact:
Supported: 100rel,timer
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: multipart/mixed,application/media_control+xml,application/sdp
Min-SE: 60
Session-Expires: 1800;refresher=uas
Max-Forwards: 9
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 317
v=0
o=BroadWorks 18275729 1 IN IP4 203.161.164.69
s=-
c=IN IP4 203.161.164.69
t=0 0
m=audio 18128 RTP/AVP 18 8 0 101
c=IN IP4 203.161.164.69
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=bsoft: 1 image udptl t38
Sep 9 10:21:41.508: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Sep 9 10:21:41.516: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=21
Content-Length: 0
Sep 9 10:21:41.544: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
CSeq: 633854439 ACK
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
Max-Forwards: 9
Content-Length: 0
Voice Config
Router#
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server expires max 3600 min 3600
localhost dns:mel.byo.engin.com.au
no call service stop
voice class codec 1
codec preference 1 g711ulaw
voice translation-rule 10
rule 1 /^0/ //
voice translation-rule 11
rule 1 /^.*/ /0386821234/
voice translation-profile PSTN_Outgoing
translate calling 11
voice-card 0
dsp services dspfarm
mgcp profile default
sccp local Vlan100
sccp ccm 10.1.100.1 identifier 1 version 7.0
sccp
sccp ccm group 1
bind interface Vlan100
associate ccm 1 priority 1
associate profile 1 register confdsp
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 4
associate application SCCP
dial-peer voice 99 voip
translation-profile outgoing PSTN_Outgoing
destination-pattern .T
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 100 voip
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number 0386821234
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 110 voip
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number .%
dtmf-relay rtp-nte
no vad
dial-peer voice 90 voip
description Melbourne 03 Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [89].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 91 voip
description National Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 0[278]........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 92 voip
description Vic/Tas 03 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [56].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 93 voip
description Mobile numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 04........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 94 voip
description 13XXXX numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 13[1-9]...
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 96 voip
description 1300/1800 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 1[38]00......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 98 voip
description Emergency 000
translation-profile outgoing PSTN_Outgoing
destination-pattern 000
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
sip-ua
credentials username 0386821234 password 7 XXXX realm voice.mibroadband.com.au
authentication username 0386821234 password 7 XXXX
nat symmetric role active
nat symmetric check-media-src
no remote-party-id
retry invite 2
retry register 10
timers connect 100
mwi-server dns:mel.byo.engin.com.au expires 3600 port 5060 transport udp unsolicited
registrar dns:mel.byo.engin.com.au expires 3600
sip-server dns:mel.byo.engin.com.au
connection-reuse
telephony-service
sdspfarm conference mute-on #1 mute-off #2
sdspfarm units 2
sdspfarm tag 1 confdsp
conference hardware
max-ephones 42
max-dn 144
ip source-address 10.1.100.1 port 2000
calling-number initiator
service phone videoCapability 1
service phone displayOnDuration 00:01
service phone displayOnTime 08:30
service phone displayOffTime 17:30
service phone displayIdleTimeout 00:01
service phone displayOnWhenIncomingCall 1
system message Cisco CME
load 7941 SCCP41.8-4-2S
load 7942 SCCP42.8-4-2S
load 7945 SCCP45.8-4-2S
load 7961 SCCP41.8-4-2S
load 7962 SCCP42.8-4-2S
load 7965 SCCP45.8-4-2S
load ata ATA030204SCCP090202A
time-zone 48
date-format dd-mm-yy
voicemail 90125200
mwi relay
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
moh music-on-hold.au
web admin system name cisco secret 5 $1$d8/H$glhLiCCWXmFSUp6BtwGho0
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 0.T
create cnf-files version-stamp 7960 Jul 06 2011 10:32:45
ephone-dn 1 dual-line
number 038682XXXX
label 101
name 7965
mwi sip
ephone-dn 2 dual-line
number 102
label 102
name 7941
ephone-dn 3 dual-line
number 103
label 103
name 7920
ephone 1
device-security-mode none
video
mac-address 0023.5EB8.6E4E
type 7965
button 1:2 2:1
ephone 3
device-security-mode none
mac-address 0019.0633.A933
max-calls-per-button 2
type 7920
button 1:3
ephone 10
device-security-mode none
mac-address 0019.E7B7.BAB3
max-calls-per-button 2
type ata
button 1:1 -
SIP trunking between Microsoft OCS server and Cisco Voice GW router.
Hello All,
I have a client with an existing Microsoft OCS (office communications server) environment with the OCS server in their head office. The OCS clients in the remote Office registers with the OCS server in the head office. The WAN connectivity between the remote office and the Head office is MPLS.I would like to facilitate local call (PSTN) features at the remote site through a newly proposed Voice gateway router.
Can I achieve this by doing a SIP trunk between the OCS server in the head office to the newly proposed voice GW router in the remote office through the existing MPLS link. If yes, Could any one please assist me in this regards or suggest any other best solution to achieve the same.
Thank you in advance,
Mohammed Ameen RHi David,
this is a normal behaviour. To CUCM, OCS is a remote destination (just like your mobile phone). When your mobile phone hangs up, the system will put the call on hold for 10 sec.
This is there for the mobile user to go to his desk to pick up the call and continue the conversation (part of single number reach feature)
The best practise will be for the user to ensure that the other party hangs up the call first before he hang up.
Please grade if you think it's useful =) -
UC520 Resets every time when making a SIP trunk Call.
I have a UC520 and i just upgraded to the 8.0.2 software pack. I also added SIP for Skype SIP trunk - when i make a call the router resets. When i do a show ver is see this line - System returned to ROM by error - a SegV exception.
never heard of that...thats not cool...open a case asap.
-
CUCM route calls diferents gateways/sip trunks
Hi at all, I have CUCM 6.1.1 and I want to route calls throughs diferents gateways or sip trunks.
I planned to do with route groups, but I can not add on a route group a H323 gateway and a SIP trunk at the same time.
How can route calls in different ways?
In the CUCM page "Route patterns" I want to make alternative routes, for example, the number 6666 is on route "666X" through a "gateway/route list", but if I can not contact by going this route I need to go through the alternative route "XXXX" through another "gateway/route list".
How can I make by going first to one pattern and then the other pattern?
Thanks!
FranOk thanks but one question more.... if I have a MGCP Gateway? Can I do this from my MGCP Gateway? or I need an H323 Gateway.
And another possibility.... I dont know if it's right....
Can I do this with Partitions and CSS?
This is for example I 'll have a CSS "Global" with Partitions (Primary and Secondary);
It could go the route first to 666x Gateway with CSS "Global" and partitions (Primary and Secondary). This way I do not know if it is routed first through the Gateway of the partition as Primary and Secondary alternative partition that is served by the SIP Trunk.
Using the "Dial Number Analyzer" I get the second path XXXX (SIP Trunk) as an alternative route ... -
Video only enabled when call is initiated from one direction across SIP Trunk
wonder If anyone can shed some light on this.
I have an issue between two cucm clusters, tied together with a SIP trunk.
If we dial from Australia to the US there is two way video and audio. If the US calls Australia, there is only audio. I have run a test call from the US through VLT and have found the following SDP's (see below). When The US make a video enabled call to australia the message "Video is not available, Remote party has video off" on the US phone screen.
Both clusters have the SIP trunk set up with the same codec settings and video bandwidth between reqions and locations. the SIP trunk is configured pretty much stock standard and identical at both ends, yet the SDP seem to want to negotiate different Video Parameters (again see SDP's below). CUCM in australia is 10.61.2.82.
what other settings can I check to get video to work when calls get initiated from either direction,...................
both phones are SIP 8941's, again audio is no problem in both directions.
=======this is from the phone in Australia to the CUCM in australia phone IP 10.61.4.112======================================
45870304.002 |09:02:07.941 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.61.4.112 on port 34271 index 53563 with 2089 bytes:
[344530309,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.61.2.82:5060;branch=z9hG4bKe0103892bbb75
From: "Anonymous" <sip:[email protected]>;tag=109791678~1b5af941-cea2-4a00-a0bd-15a532224d7d-59374526
To: <sip:[email protected]>;tag=5057a887bfdd550c0d321a20-7f843426
Call-ID: [email protected]
Date: Wed, 29 Apr 2015 23:02:07 GMT
CSeq: 101 INVITE
Server: Cisco-CP8941/9.4.2
Contact: <sip:[email protected]:34271;transport=tcp>;video
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Dennis Mink - 33935" <sip:[email protected]>;party=called;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Content-Length: 966
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 28123 0 IN IP4 10.61.4.112
s=SIP Call
t=0 0
m=audio 16736 RTP/AVP 0 8 18 102 9 116 101
c=IN IP4 10.61.4.112
a=trafficclass:conversational.audio.avconf.aq:admitted
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 16738 RTP/AVP 126 97
c=IN IP4 10.61.4.112
b=TIAS:2000000
a=trafficclass:conversational.video.avconf.aq:admitted <----this is missing from US SDP
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428014;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200;max-rcmd-nalu-size=1300
a=imageattr:126 send * recv [x=640,y=480]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428014;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200
a=imageattr:97 send * recv [x=640,y=480]
a=rtcp-fb:* ccm tmmbr
a=sendrecv
============below is coming from the US (phone IP is 10.1.109.81)================
04/30/2015 09:02:08.169 Send 10.61.4.112 SIP ACK bfa99a00-541162ed-71da57-52023d0a NotAvail
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.61.4.112 on port 34271 index 53563
[344530326,NET]
ACK sip:[email protected]:34271;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.61.2.82:5060;branch=z9hG4bKe010481f320b08
From: "Anonymous" <sip:[email protected]>;tag=109791678~1b5af941-cea2-4a00-a0bd-15a532224d7d-59374526
To: <sip:[email protected]>;tag=5057a887bfdd550c0d321a20-7f843426
Date: Wed, 29 Apr 2015 23:02:05 GMT
Call-ID: [email protected]
User-Agent: Cisco-CUCM10.0
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 456
SDP Message
====================================================
v=0
o=CiscoSystemsCCM-SIP 109791678 1 IN IP4 10.61.2.82
s=SIP Call
c=IN IP4 10.1.109.81
b=TIAS:8000
b=AS:8
t=0 0
m=audio 16412 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=trafficclass:conversational.audio.aq:admitted <---what does this do here, and how?
m=video 0 RTP/SAVP 31 34 96 97 <-----------port 0. why?
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:96 H263-1998/90000
a=rtpmap:97 H264/90000
a=content:main
a=inactiveHi Dennis,
On US phone SDP media attribute is inactive.
a=rtpmap:97 H264/90000
a=content:main
a=inactive
Are you sure that audio works ? Can you please share all the SIP messages of both the scenarios.
Thanks
Manish -
Adding ICT trunk and SIP trunk into Route group
Hi ,
We need to map ICT and SIP trunk into the same route group ,but the problem here is already same ICT is mapped to another route pattern.
If i try to create new ICT with same remote IP ,it's throwing Add failed because the remote IP is already defined.
Is there anyway we can add ICT with same remote IP and map the ICT and SIP trunk ? or Is there anyway that we can add exisitng ICT into route pattern.
Route pattern is used for this route group is different.CUCM version - 7.X.Please advice.
Regards,
RamanathanThanks Suresh ...
In that case ,I can assign the route group(ICT ,SIP - Top Down) to two different Route pattern.
Both patterns will hit ICT first ,Please correct me if I am wrong.
Ram -
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. -
Problems between an UC520 and Asterisk with sip trunk
I have an UC520 and Asterisk with a sip trunk created between them, the calls from the UC520 to the Asterisk are ok, but the calls form de Asterisk to the UC520 are always busy.
Logs from the asterisk show that the first part of the call is ok, but the call is not complete, this means that the part where the extensions are with @ipuc520 doesn't appear
I created a sip trunk from de CCA 1.9 and it puts this for incoming calls for the dial peer, if I compare with a CCME, there is no configuration for incoming call there
/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Tabla normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
mso-para-margin:0cm;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:Calibri;
mso-fareast-theme-font:minor-latin;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;
mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;
mso-fareast-language:EN-US;}
dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:x.y.z.w
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
And there is no configurarion at all that could block the calls
The x.y.z.w was the sip server ip (asterisk ip)
The comminication between sip and h323 are allowed in the four ways
The allowed codecs are g711ulaw and g729r8
Asterisk is working now with other CCME and they are ok so I copied the configuration from those CCME to the UC520 and from the other sip trunks in asterisk the new trunk sip for uc520
The sip trunk created from the CCA was replaces for the one from the CCME that is working now
The routes are ok in Asterisk.
There is no translation profile in incoming calls.
There is no ACL applied in all configuration.
There is no log about callres incoming from the asterisk.
Could anyone halp me pls?Hi Rina,
Help me to try and understand what you are trying to do.
In this code snippet i see the following:
001808: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=7129, Called Number=7129, Peer Info Type=DIALPEER_INFO_SPEECH
001809: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=7129
001810: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
001811: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20036
This looks as though you have a call coming in from the Asterisk system to number 7129, which then leads to this according to the config file you provided.
number 7129
label 7129
description7129
name 7129
call-forward busy 6001
call-forward noan 6001 timeout 10
Which at this point I am going to assume this is ephone-dn 10 (Please confirm). If this is the case then the inbound call is being matched correctly to a DN (Which has its own dial-peer tag "Dial-peer Tag=20036".
But then i see this:
001817: 1w3d: //-1/55940098BA19/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1000
001818: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=Unknown, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
001819: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
So the incoming call has been matched to Dial-peer 1000 which is an incoming VoIP dial-peer:
dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
But then can see it has no where to go. So either I am reading this all wrong and the 7129 number is a result of another call taking place whilst you were debugging the system, or it is part of the debug and I am missing something here.
Rina, just so I understand this all. Are you trying to do WAN type calling from one system UC-500 (System "A") to the Asterisk system ( System B) and same? And so far calls going from the UC-500 to the Asterisk system are fine, but calls coming in from the Asterisk system to the UC-500 are not?
What happens on the Asterisk side when you try to call an Extension on the UC-500, do you get any ringing? Or is it a fast busy tone?
I am going to look over your configuration and debug a little further when I get home, maybe I am missing something here and can identify it.
Cheers,
David. -
Hello!
Now we use Cisco IP telephony based on Cisco CME. Cisco CME is installed on 2821 router with IOS C2800NM-ADVENTERPRISEK9-M, Version 12.4(24)T7. A SIP trunk is configured between CME router and telephony provider.
We plan to upgrade our telephone system and begin using Cisco CBE6000. I study the documentation and in "Cisco Business Edition Unified Communications and Collaboration Solutions for Small and Midsized Companies", I find that for SIP trunk to works in CBE6000 a Cisco Integrated Services Router with Cisco Unified Border Element is required. Or other quote from this document "Public switched telephone network (PSTN) connections using SIP is available through any Cisco Integrated Services Router voice gateway".
In this regard there is a question: what is the functions of this router when I make a trunk between CBE6000 and telephony provider SIP gateway and whether it is possible to use existing Cisco 2821 router as such router.
Thanks.Hi Ayodeji,
I too have a somewhat similar query.
We have UC560 running currently at our company, the UC560 is connected to an ISDN modem from the network provider. We are now upgrading to BE6000. To connect to the ISDN E1/T1 line, we've ordered Cisco 2921 with a E1/T1 VWIC card (VWIC3-2MFT-T1/E1) which will connect to BE6000 and internal LAN. I just wanted to know what would I need to configure on 2921?
Do I need to install CUBE or just configure it to connect to ISDN line?
Many thanks for any help.
regards,
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