CallManager 4.x to SIP ITSP
Hello,
I am trying to find out if a CallManager 4.x system can communicate to an IP Telephony Service Provider over SIP to their Sessions Border Controller. From what it looks like, I will need an IP2IP gateway to talk H323 to the CCM and SIP to the ITSP. Has anyone successfully done this before?
Any help or experiences would be greatly appreciated.
We have tested this in our lab, and this was working well. An Cisco 2811 with ver. 12.4 IP2IP was used for this test and H323 to SIP, H323 to H323 and SIP to H323 was working well.
Config Cisco router :
Building configuration...
Current configuration : 2983 bytes
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
enable password cisco
no aaa new-model
ip cef
no ip dhcp use vrf connected
no ip dhcp conflict logging
ip dhcp excluded-address 10.193.25.1 10.193.25.65
ip dhcp excluded-address 172.16.1.1 172.16.1.9
ip dhcp excluded-address 10.193.25.70 10.193.25.80
ip dhcp pool 10.193.25.0
network 10.193.25.0 255.255.255.0
option 150 ip 10.193.25.113
default-router 10.193.25.111
lease infinite
multilink bundle-name authenticated
voice-card 0
no dspfarm
dsp services dspfarm
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
call start interwork
sip
interface Loopback0
ip address 10.0.0.1 255.255.255.255
interface FastEthernet0/0
ip address 10.193.25.111 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 10.193.25.111
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Content-Engine1/0
no ip address
shutdown
ip route 0.0.0.0 0.0.0.0 10.193.25.1
no ip http server
control-plane
dial-peer voice 10 voip
description VoIP to live callmanager
destination-pattern 3...
progress_ind connect enable 8
session target ipv4:10.193.1.5
dtmf-relay h245-alphanumeric
codec g711alaw
dial-peer voice 20 voip
description VoIP to Test Callmanager
tone ringback alert-no-PI
destination-pattern 2...
progress_ind setup enable 3
progress_ind progress enable 8
progress_ind connect enable 8
session target ipv4:10.193.25.113
dtmf-relay h245-alphanumeric
codec g711alaw
dial-peer voice 30 voip
description to VoIP/AA at Test Callmanager
destination-pattern 500.
session target ipv4:10.193.25.113
dtmf-relay h245-alphanumeric
codec g711alaw
dial-peer voice 1 voip
description to H323 External GW
destination-pattern 0T
session target ipv4:10.193.1.4
dtmf-relay h245-alphanumeric
codec g711alaw
dial-peer voice 200 voip
description to SIP Soft IP-Phone
destination-pattern 1999
session protocol sipv2
session target ipv4:10.193.10.9
dtmf-relay rtp-nte
codec g711alaw
dial-peer voice 100 voip
tone ringback alert-no-PI
description 3th party hardware SIP IPPhone
destination-pattern 1...
session protocol sipv2
session target ipv4:10.193.25.200:5060
dtmf-relay rtp-nte h245-alphanumeric
codec g711alaw
no vad
sip-ua
retry options 0
gatekeeper
shutdown
line con 0
line aux 0
line 66
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120
line vty 0 4
password cisco
login
scheduler allocate 20000 1000
end
Router#
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SIP ITSP on CUCM 10.5.2 (No CUBE) Incoming calls fail, outgoing are fine
Hi,
I am in the process of upgrading a customer who is on 8.0.3. They have an ITSP terminating SIP Trunk directly on the CCM Server
I upgraded the system to 10.5.2. During cutover I was able to make outgoing calls but all incoming calls were failing.
After reverting back to the old system, everything is working fine again, and I dont understand what could be the possible issue that it doesnt work on 10.5.2 but it works well on 8.0.3.
I checked almost everything and dont find anything that stands out, which may be contributing to the issue.
Any idea what could be missing here?
ThanksThanks for all your tips.
It was turned out that, the URI was a FQDN and during the first install of the 8.0.3 (in the sandbox) I had not bothered to get the DNS Services replicated and then didnt check if the ITSP was sending the invite on URI based on FQDN or IP Address
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Hello all,
I have a situation where I am trying to connect a CME 3.2 system, running on a 2821, to Global Crossing' s Local Dial and Outbound Dial service. They are using an Acme Session Border Controller, which at this point, does not support "SIP Redirect". Basically, when we forward our phones to an outbound #, the CME sends a SIP message telling the Acme Session Border Controller to redirect the call to an outbound port on the Acme.
I'm wondering if there is a way to add a router to the mix on my side as an IP2IP gateway and connect the CME to this gateway using H.323 and have the IP2IP gateway initiate another call to the Acme, thereby getting around this SIP redirect issue. Does anyone know if this might work?
Thanks, as always, for any assistance!
-DarinWayne,
I did get some features to work, however, there were serveral querks that ultimately convinced me to stay away from the solution until it matures. In my scenario, I did not need a userid and pin as Global Crossing authenticates you by putting you on a VRF in their MPLS network. I'm told that Cisco will release call manager 5.x this fall, which will natively support SIP to the instrument. I have a plan to re-evaluate this when that product comes out and is showing signs of stability.
HTH,
Darin -
Prefixing a 9 and 91 to incoming calls from SIP provider for callback
I am wondering what would be the best options for prefixing a 9 or 91 to incoming calls over a sip connection to allow callback from missed calls and recieved calls. The setup is
callmanager 7.1.5 >>>>sip trunk>>>>>>>>>>>>CUBE>>>>>>>>>>>sip to ITSP
I am thinking voice translation rules is the only option for this? any configuration examples for this would be greatly appreciated.
would this work?
voice-translation rule 1
rule 1 // /9/
voice-translation profile prefix_9
translate calling 1
dial-peer voice 101 voip
destination-pattern ???????...$
voice-class codec 1
session protocol sipv2
session target ipv4: to callmanager
incoming called-number .
dtmf-relay rtp-nte
dial-peer voice 1001 voip
translation profile incoming prefix_9
destination-pattern T
session protocol sipv2
session target ipv4: to sip provider
incoming called-number ???????...$
dtmf-relay rtp-nteYour config should work fine, except your profile is only applied to one dial-peer, make sure you apply it to the one that is used to redirect the call to CUCM.
Also, you did not mention what country you are in, but if this is US you may want to prefix 91 to national calls as carriers don't provide 9 as part of the CLID delivery, also what about your international calls, you may would be more explicit in your first rule to match for national digit string and then have another rule for international.
HTH,
Chris -
i have a few sip/h323 providers. I have also enabled sip/h323 on my as5400xm(this is for my asterisk server). Since i'm using these providers, i have to put their IP in my access-list. my concern is, since my gateway is accepting sip/h323 calls. what if these provider send the calls to my gateway? so i was thinking of a way to restrict this. It could be as simple as tweaking the access-list. but I don't know. Please help.
here's how i have my access-list setup:
access-list 101 permit tcp host 10.10.10.10 any
access-list 101 permit udp host 10.10.10.10 any
access-list 101 permit udp any any range 16384 32767
access-list 101 deny tcp any any
access-list 101 deny udp any any
Thanks in advanceAh, so you just want to restrict VoIP calls from L3 addresses other than your provider?
That's just a simple ACL to open up traffic to your SIP ITSP's IP external addresses, and block anything else.
You can get what IPs and ports are used by your provider, but here is what you need open on the Cisco side inbound for an inbound ACL on a WAN interface:
UDP - ITSP address:ITSP SIP Port to External interface:5060 - For SIP signaling
ITSP address:ITSP RTP Port Range - External interface:16384-32767 - RTP traffic
ITSP's port range could be anything between 1024-65535. SIP usually comes from UDP/5060 from the ITSP, but doesn't have to. Verify with them, or look at a SIP debug or packet capture to verify.
The implicit deny will take care of everything else. -
SIP trunk call to CTI port media endpoint problem
Hi, I have the following scenario:
CUCM ---SIP-----CUBE----SIP----ITSP
CUCM cluster has two servers 10.1.9.5 and 10.1.9.6 (subscriber), we have a CTI application running on 10.1.9.8, we got a bunch of SIP DIDs from ITSP and those SIP DID numbers are mapped to CUCM internal DNs on CUBE through translation, inbound SIP calls to mapped SCCP phones work fine, media stream of internal call leg is established directly between SCCP phone and CUBE as expected. However, when inbound SIP calls to the number which maps to CTI extention, there is no audio, debug on CUBE shows that the media stream of internal call leg is between one of the CUCM servers. I want to understand why CUCM is telling CUBE to use the IP address which is NOT CTI port is registered from (in this case 10.1.9.8)? why would CUCM treat a CTI extention differently from a regular SCCP phone extension. For regular inbound call to SCCP phones codec between CUCM-CUBE and between CUBE-ITSP are both g11ulaw.
Thanks,"debug on CUBE shows that the media stream of internal call leg is between one of the CUCM servers" ==>
debug on CUBE shows that the media stream of internal call leg is between 10.1.9.5 (happens to be MTP) and CUBE -
Hi
We are migrating from Analogue to IP Telephony. I have recieved the following guidlines to configure the SIP Trunk:
*For signaling: use IP : x.x.211.70 ( SIP ) on PORT 5060
*Regarding Numbering Format, use the following:
• For outgoing Calls :
The originating Number (A#), should be 96611510XXXX format.
The Destination Number should be 0NXXXXXX (N area code) or 00XXXXXXXXX (for international)
• For incoming Calls:
The Destination Number (B#), should 011510XXXX Format.
The originating Number (A#), will be 0NXXXXXXX or 00XXXXXXXXXXX Format
*Use Audio Codec's G711-aLaw ; G711-uLaw & G729
*Use T.38 For FAX
*set DTMF to RFC2833
*Make sure to reply with 200Ok for our OPTIONS messages ( ping messages for the SIP)
* configure the following SIP Timers: “Min-SE=1800 “’ & “Expires=300”
For connectivity consider the following:
SIP CE: 10.65.13.110 (it might be needed to translate this IP to the PBX local IP).
SIP GW: 10.65.13.109
Subnet mask: /30
SIP VLAN: 1191
Notes:
Kindly make sure to have GO SIP GW (x.x.211.70) routed to SIP GW (10.65.13.109) as next-hop.
Kindly make sure to have SIP CE IP addresses are in VLAN 1191.
Can please anyone explain what have to done?
RegardsAhmed,
Wao..Where do I start...This information is required for configuration on your CUBE..which will be your 2921 router...
Ahmed, here are some pointers I wrote a while ago..
In addition to these points, you will need to configure your cube to be able to route traffic to your ITSP using all the information given to you
1. Configure CUBE for media flow through. In this Mode CUBE acts as a true B2BUA. Advantages you get include address hiding and security becaue CUBE terminates and re-originate both signalling and Media. In this mode CUBE becomes a point of demarcation from th external world.
2. Configure CUCM to use Delayed Offer and CUBE to convert delayed offer to ealry offer...This prevents the need for you to use MTP to send Early offer on CUCM
voice service voip
early-offer forced
3. Configure DTMF signalling method on sip trunk to "No preference" This setting allows Unified CM to make an optimal decision for DTMF and to minimize MTP allocation.
4. Configure your CUBE to meet the requirements of your ITSP. Ask if they have configuration templates or any specific configuration they like you to use. This will save you time troubleshooting. Most of them dont use the default port 5060 because of security, confirm with your proivider what ports they use.
voice service voip
allow-connections sip to sip
sip
early-offer forced
header-passing
error-passthru
5. Use SIP to SIP...Use end to end sip. CUCM---sip---CUBE--sip----ITSP
6. Create a Trusted list of IP addresses on your CUBE is your CUBE IOS is 15.1 .2(T) and above.
voice service voip
ip address trusted list
ipv4 203.0.113.100 255.255.255.255
ipv4 192.0.2.0 255.255.255.0
This is imprtant because sometimes your ITSP will send you a single ip address for signalling and will then send media on a different IP adress. So get all the IP address your ITSP is using and add them to the trust list as shown above
7. Configure your inbound and outbound dial-peer approriately
Inbound Dial-Peer for calls from CUCM to CUBE (CUCM sending 9 +all digits dialled to CUBE)
dial-peer voice 100 voip
description *** Inbound LAN side dial-peer ***
incoming called-number 9T
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to CUCM (SP will be sending 10 digits inbound)
dial-peer voice 200 voip
description *** Outbound LAN side dial-peer ***
destination-pattern [2-9].........
session protocol sipv2
session target ipv4:
codec g711ulaw
dtmf-relay rtp-nte
Note: If more than 1 CUCM cluster exists, you will have to create multiple such LAN dial-peers with “preference CLI” for CUCM redundancy/load balancing
Inbound Dial-Peer for calls from SP to CUBE
dial-peer voice 100 voip
description *** Inbound WAN side dial-peer ***------------------(catch-all for all inbound PSTN calls)
incoming called-number [2-9].........
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to SP
dial-peer voice 200 voip
description *** Outbound WAN side dial-peer ***
translation-profile outgoing Digitstrip
destination-pattern 9[2-9].........
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
session target ipv4::XXXX (where XXXX is the port number your provider is using if different from 5060)
codec g711ulaw
dtmf-relay rtp-nte
8. SIP Normalization:
You may need to configure sip normnalization to modify sip headers, CLI etc. A good example is during call forwarding. You may need to change your diversion headers to match the CLI your provider is expecting.. if your call forwarding is failing during testing this may be the reason..We can help you with this.
9. Media Resources
Plan your solution properly. Consider if you will need Xcoders, MTP, Conference bridge etc. You may avoid the need for xcoders if you confure your regions properly and use voice class codecs on your sip profiles. It is important to know if there are any endpoints in your network that do not support dtmf relay rtp-nte. You can avoid the use of MTP if you configure your dial-peers to have multiple dtmf types for thos phones that do not support rtp-nte
e.g
dial-peer voice 1 voip
session protocol sipv2
dtmf-relay rtp-nte digit-drop sip-kpml (if your phones support kpml..then this will be used)
If in your environment you will need to do xcoding or CFB then ensure you have PVDMS
.10.FAX
If you have FAX in your network, determine what fax protocol your sip provider supports. Dont assume. Ask them and confirm in writing what they support. I have seen legal cases because of fax failures over sip trunks
Configure your FAX devices in seperate device pools and use porefix to route calls using G711 only. Even if you are using T38, ensure your fax use G711 to establish the voice calls
Finally
11. Have a detailed and carefully planned TEST Plan. Test the FF:
Inbound and outbound Local, Long distance, International calls for G711 & G729 codecs (if supported by provider)
Outbound calls to information and emergency services
Caller ID and Calling Name Presentation
Supplementary services like Call Hold, Resume, Call Forward & Transfer
DTMF Tests
Fax calls – T.38, modem pass-through--whichever one you decide to use
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared" -
Hello
I have 2 sites that the phone is working via SIP, and we are having the issue when you call from one site to another the first calls are getting busy tone.
I checked the bandwidth between sites and it's ok, what other tests can I do to check and find where can the issue be.
RegardsOk I may be missing something here, let's start by understanding your call flow within your Branches, you have both extensions registered to Call manager, correct? Can you validate that they are indeed registered with a line number? Then, what's the transport between the 2 sites? MPLS? You said you have SIP but you certainly wouldnt need a SIP trunk to call between extensions in the same Call manager, your SIP trunk would be needed for PSTN access. Please give us an idea of your call flow, is it this?:
Phone -> CUCM ->SIP-CUBE->SIP-ITSP?
Thanks,
Frank -
CALL DOES NOT ROUTE OUT THE LOCAL GATEWAY
Local calls will not route out the local Gateway of branch1 to the PSTN or from the PSTN back to branch1, however they will route out either CorpHQ or branch2 backup gateways. When I go into the route group configuration for branch1, and remove the backup gateways, I get a fast busy tone when I dial the local number. I know the MGCP Gateway at branch1 is functioning because when I dial 911 and run debug ISDN Q931, the call routes properly through branch1, so I have a call routing problem. I ran DNA and it came back as ROUTE THIS PATTERN and all of the number translations looked accurate, so I didn't have to check for any block patterns. I'm not getting any errors on the calling party phone display. When I deleted the route pattern for the branch1 site and forced it to use the global route pattern, I received a debug output on branch1. I do not know a debug command (such as debug voip dial-peer or debug ccsip messages) to use for an MGCP Gateway to see if the call is actually reaching the Gateway.
I have checked the following:
the route pattern configuration
the translation pattern configuration
the called party transformation pattern configuration
the route list configuration to make sure the correct route group for branch1 was selected
the route group configuration to make sure that the branch1 Gateway was first in the order of selected devices
the route pattern configuration to make sure the correct route list for branch1 ist selected
the Gateway configuration to make sure it's using the device pool for branch1 and to make sure the called party transformation CSS for the branch1 Gateway is applied
the device pool configuration to make sure it's using the route group branch1
Any assistance would be greatly appreciated
Regards,
RonHi Nishant:
Please see the attachments for the Gateway pages
The significant digits for inbound calls for all 3 gateways is '4'
Please see the running-configs of the 3 gateways and the PSTN
Please see the debugs for the INBOUND calls
Many Thanks,
Ron
The following INBOUND call from the PSTN to 2065011001 is now working, however it is supposed to be routing through CorpHQ and is instead routing through Branch1. Please see 'DEBUG VOIP CCAPI INOUT' & 'DEBUG ISDN Q931'
Branch1#
ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x0096
Cause i = 0x8290 - Normal call clearing
//22/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x16, digit_event=0x0, enable=FALSE, consume=FALSE)
//22/5A001212800B/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=22
//22/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x49E07FD4, callID=0x16, disp=0, digit_event=0x0, enable=FALSE, consume=FALSE)
//22/5A001212800B/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0x0, Interface=0x49E07FD4, Call Id=22
//22/5A001212800B/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=4000(ms), Inter Digit Timeout=4000(ms))
//22/5A001212800B/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Network, Params=0x0, Call Id=22
//23/5A001212800B/CCAPI/ccGetCallStatistics:
Call Stats=0x4A5346FC, Call Id=23
//22/5A001212800B/CCAPI/ccConferenceDestroy:
Conference Id=0xC, Tag=0x0
//22/xxxxxxxxxxxx/CCAPI/cc_api_bridge_drop_done:
Conference Id=0xC, Source Interface=0x49E07FD4, Source Call Id=22,
Destination Call Id=23, Disposition=0x0, Tag=0x0
//23/xxxxxxxxxxxx/CCAPI/cc_api_bridge_drop_done:
Conference Id=0xC, Source Interface=0x495BABA4, Source Call Id=23,
Destination Call Id=22, Disposition=0x0, Tag=0x0
//22/5A001212800B/CCAPI/cc_generic_bridge_done:
Conference Id=0xC, Source Interface=0x495BABA4, Source Call Id=23,
Destination Call Id=22, Disposition=0x0, Tag=0x0
//22/5A001212800B/CCAPI/ccCallDisconnect:
Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
//22/5A001212800B/CCAPI/ccCallDisconnect:
Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
//22/5A001212800B/CCAPI/cc_api_get_transfer_info:
Transfer Number Is Null
//23/5A001212800B/CCAPI/ccCallDisconnect:
Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
//23/5A001212800B/CCAPI/ccCallDisconnect:
Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
//23/5A001212800B/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x495BABA4, Tag=0x0, Call Id=23,
Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
//23/5A001212800B/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
:cc_free_feature_vsa freeing 4821DDE8
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
vsacount in free is 1
//22/5A001212800B/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x49E07FD4, Tag=0x0, Call Id=22,
Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
//22/5A001212800B/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
:cc_free_feature_vsa freeing 4821DEC8
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
vsacount in free is 0
ISDN Se0/0/0:23 Q931: TX -> RELEASE pd = 8 callref = 0x8096
ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0096
ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x0097
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA18381
Preferred, Channel 1
Progress Ind i = 0x8183 - Origination address is non-ISDN
Display i = 'Seattle US Phone'
Calling Party Number i = 0x4180, '2065015111'
Plan:ISDN, Type:Subscriber(local)
Called Party Number i = 0xC1, '2065011001'
Plan:ISDN, Type:Subscriber(local)
//-1/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x49E07FD4, Interface Type=6, Destination=, Mode=0x9,
Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=, FinalDestinationFlag=FALSE, Outgoing Dial-peer=0, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=D000000002f5368f000000F580000097)
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:cc_get_feature_vsa malloc success
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
cc_get_feature_vsa count is 1
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:FEATURE_VSA attributes are: feature_name:0,feature_time:1210179280,feature_id:24
//24/74820328800C/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=6, FlowMode=1
//24/74820328800C/CCAPI/ccCallSetContext:
Context=0x4A524790
//-1/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x495BABA4, Interface Type=9, Destination=0.0.0.0, Mode=0x9,
Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=, FinalDestinationFlag=FALSE, Outgoing Dial-peer=0, Call Count On=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=D000000002f5368f000000F580000097)
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:cc_get_feature_vsa malloc success
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
cc_get_feature_vsa count is 2
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:FEATURE_VSA attributes are: feature_name:0,feature_time:1210179056,feature_id:25
//25/74820328800C/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=9, FlowMode=1
//25/74820328800C/CCAPI/ccCallSetContext:
Context=0x4A524580
//25/74820328800C/CCAPI/cc_api_call_connected:
Interface=0x495BABA4, Data Bitmask=0x0, Progress Indication=NULL(0),
Connection Handle=0
//25/74820328800C/CCAPI/cc_api_call_connected:
Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
//24/74820328800C/CCAPI/cc_api_call_proceeding:
Interface=0x49E07FD4, Progress Indication=NULL(0)
//24/74820328800C/CCAPI/cc_api_call_connected:
Interface=0x49E07FD4, Data Bitmask=0x1, Progress Indication=DESTINATION IS NON ISDN(2),
Connection Handle=0
//24/74820328800C/CCAPI/cc_api_call_connected:
Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
//24/74820328800C/CCAPI/ccCallModify:
Nominator=0x1000, Params=0x4A2E7368, Call Id=24
//24/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x18, digit_event=0x1, enable=TRUE, consume=FALSE)
//24/74820328800C/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=24
//24/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x49E07FD4, callID=0x18, disp=0, digit_event=0x1, enable=TRUE, consume=FALSE)
//24/74820328800C/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0x0, Interface=0x49E07FD4, Call Id=24
//24/74820328800C/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=15000(ms), Inter Digit Timeout=10000(ms))
//24/xxxxxxxxxxxx/CCAPI/ccConferenceCreate:
(confID=0x4A2E757C, callID1=0x18, callID2=0x19, tag=0x0)
//24/xxxxxxxxxxxx/CCAPI/ccConferenceCreate:
(confID=0x4A2E757C, callID1=0x18, gcid=0-0-0-0, tag=0x0)
//25/xxxxxxxxxxxx/CCAPI/ccConferenceCreate:
(confID=0x4A2E757C, callID2=0x19, gcid=0-0-0-0, tag=0x0)
//24/74820328800C/CCAPI/ccConferenceCreate:
Conference Id=0x4A2E757C, Call Id1=24, Call Id2=25, Tag=0x0
//24/xxxxxxxxxxxx/CCAPI/cc_api_bridge_done:
Conference Id=0xD, Source Interface=0x49E07FD4, Source Call Id=24,
Destination Call Id=25, Disposition=0x0, Tag=0xFFFFFFFF
//25/xxxxxxxxxxxx/CCAPI/cc_api_bridge_done:
Conference Id=0xD, Source Interface=0x495BABA4, Source Call Id=25,
Destination Call Id=24, Disposition=0x0, Tag=0x0
//24/74820328800C/CCAPI/cc_generic_bridge_done:
Conference Id=0xD, Source Interface=0x495BABA4, Source Call Id=25,
Destination Call Id=24, Disposition=0x0, Tag=0x0
//24/74820328800C/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0xD, Destination Call Id=25)
//25/74820328800C/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0xD, Destination Call Id=24)
//24/74820328800C/CCAPI/cc_api_caps_ind:
Destination Interface=0x495BABA4, Destination Call Id=25, Source Call Id=24,
Caps(Codec=0x1, Fax Rate=0x1, Vad=0x1,
Modem=0x2, Codec Bytes=20, Signal Type=3)
//24/74820328800C/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
//25/74820328800C/CCAPI/cc_api_caps_ind:
Destination Interface=0x49E07FD4, Destination Call Id=24, Source Call Id=25,
Caps(Codec=0x4, Fax Rate=0x2, Vad=0x1,
Modem=0x0, Codec Bytes=20, Signal Type=2)
//25/74820328800C/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
//25/74820328800C/CCAPI/cc_api_caps_ack:
Destination Interface=0x49E07FD4, Destination Call Id=24, Source Call Id=25,
Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=OFF(0x1),
Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=9314)
//24/74820328800C/CCAPI/cc_api_caps_ack:
Destination Interface=0x495BABA4, Destination Call Id=25, Source Call Id=24,
Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=OFF(0x1),
Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=9314)
//24/74820328800C/CCAPI/cc_api_call_modify_done:
Result=0, Interface=0x49E07FD4, Call Id=24
//24/74820328800C/CCAPI/cc_api_voice_mode_event:
Call Id=24
//24/74820328800C/CCAPI/cc_api_voice_mode_event:
Call Entry(Context=0x4A524790)
//24/74820328800C/CCAPI/cc_process_notify_bridge_done:
Conference Id=0xD, Call Id1=24, Call Id2=25
//24/74820328800C/CCAPI/ccSetDigitTimeouts:
Initial Digit Timeout=4000(ms), Inter Digit Timeout=4000(ms)
//24/74820328800C/CCAPI/ccSetDigitTimeouts:
Call Entry(Inter Digit Timeout=4000(ms), Initial Digit Timeout=4000(ms))
//24/74820328800C/CCAPI/ccRestartDigitTimeoutMsec:
Digit Timeout=0, Call Id=24
//24/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x18, digit_event=0x1, enable=TRUE, consume=FALSE)
//24/74820328800C/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=24
//24/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x49E07FD4, callID=0x18, disp=0, digit_event=0x1, enable=TRUE, consume=FALSE)
//24/74820328800C/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0x0, Interface=0x49E07FD4, Call Id=24
//24/74820328800C/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=4000(ms), Inter Digit Timeout=4000(ms))
ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0x8097
Channel ID i = 0xA98381
Exclusive, Channel 1
//24/74820328800C/CCAPI/ccCallModify:
Nominator=0x1000, Params=0x4A2E6E68, Call Id=24
//24/74820328800C/CCAPI/cc_api_call_modify_done:
Result=0, Interface=0x49E07FD4, Call Id=24
ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8 callref = 0x8097
Progress Ind i = 0x8088 - In-band info or appropriate now available
//24/74820328800C/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Ring Back,
Tone Direction=Network, Params=0x0, Call Id=24
//24/74820328800C/CCAPI/cc_handle_inter_digit_timer:
Generate inter-digit timeout CC_EV_CALL_DIGIT_END event
The following INBOUND call from the PSTN to 5126022001 fails and is supposed to be routing through Branch1 and is instead routing through CorpHQ. Please see 'DEBUG VOIP CCAPI INOUT'
CorpHQ#
//-1/A31ADF52800B/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=5126026222
cisco-anitype=4
cisco-aniplan=1
cisco-anipi=0
cisco-anisi=0
dest=5126022001
cisco-desttype=4
cisco-destplan=1
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-lastrdn=
cisco-rdntype=-1
cisco-rdnplan=-1
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
//-1/A31ADF52800B/CCAPI/cc_api_call_setup_ind_common:
Interface=0x49F42894, Call Info(
Calling Number=5126026222,(Calling Name=)(TON=Subscriber, NPI=ISDN, Screening=Not Screened, Presentation=Allowed),
Called Number=5126022001(TON=Subscriber, NPI=ISDN),
Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE,
Incoming Dial-peer=1, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
//-1/A31ADF52800B/CCAPI/ccCheckClipClir:
In: Calling Number=5126026222(TON=Subscriber, NPI=ISDN, Screening=Not Screened, Presentation=Allowed)
//-1/A31ADF52800B/CCAPI/ccCheckClipClir:
Out: Calling Number=5126026222(TON=Subscriber, NPI=ISDN, Screening=Not Screened, Presentation=Allowed)
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:cc_get_feature_vsa malloc success
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
cc_get_feature_vsa count is 1
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:FEATURE_VSA attributes are: feature_name:0,feature_time:1241383960,feature_id:13
//13/A31ADF52800B/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=5126026222(TON=Subscriber, NPI=ISDN, Screening=Not Screened, Presentation=Allowed),
Called Number=5126022001(TON=Subscriber, NPI=ISDN))
//13/A31ADF52800B/CCAPI/cc_process_call_setup_ind:
Event=0x497D0010
//-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 5126022001
//13/A31ADF52800B/CCAPI/ccCallSetContext:
Context=0x4A131A54
//13/A31ADF52800B/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 13 with tag 1 to app "_ManagedAppProcess_Default"
//13/A31ADF52800B/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
//13/A31ADF52800B/CCAPI/ccCallDisconnect:
Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
//13/A31ADF52800B/CCAPI/ccCallDisconnect:
Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
//13/A31ADF52800B/CCAPI/cc_api_get_transfer_info:
Transfer Number Is Null
//13/A31ADF52800B/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x49F42894, Tag=0x0, Call Id=13,
Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)
//13/A31ADF52800B/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
:cc_free_feature_vsa freeing 49FE0410
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
vsacount in free is 0
PSTN#sh run
Building configuration...
Current configuration : 13975 bytes
! No configuration change since last restart
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
hostname PSTN
boot-start-marker
boot-end-marker
card type e1 0 0
card type t1 0 1
logging message-counter syslog
no aaa new-model
clock timezone EST -5
clock summer-time EST recurring
network-clock-participate wic 0
network-clock-participate wic 1
no network-clock-participate aim 0
dot11 syslog
ip source-route
ip cef
no ip domain lookup
ip domain name att.com
ip name-server 177.1.100.110
ip multicast-routing
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-ni
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol cisco
sip
bind control source-interface Loopback10
bind media source-interface Loopback10
header-passing
voice translation-rule 101
rule 1 /^\+.*/ //
rule 2 /^501.*/ //
rule 3 /^1206.*/ //
rule 4 /^00.*/ //
rule 5 /^0011.*/ //
rule 6 /^206/ /1206/
rule 7 /^1512.*/ /\0/
rule 8 /^011\(.*\)/ /\1/
voice translation-rule 102
rule 1 /^1\(2065015111\)$/ /\1/ type any subscriber plan any isdn
rule 2 /^1\(2065015555\)$/ /\1/ type any subscriber plan any isdn
rule 3 /^1\(2065015151\)$/ /\1/ type any subscriber plan any isdn
rule 4 /^1\(5126026222\)$/ /\1/ type any national plan any isdn
rule 5 /^31670357575$/ /&/ type any international plan any isdn
rule 6 /^31207037333$/ /&/ type any international plan any isdn
rule 7 /^31107047444$/ /&/ type any international plan any isdn
rule 8 /^911$/ /&/ type any unknown plan any unknown
rule 9 /^15126022.../ /&/ type any unknown plan any unknown
rule 10 /^31207033.../ /&/ type any unknown plan any unknown
rule 11 /^....$/ /&/ type any unknown plan any unknown
voice translation-rule 103
rule 1 /^206.*/ /&/ type any subscriber plan any isdn
rule 2 /^1/ // type any national plan any isdn
rule 3 /^00/ // type any international plan any isdn
voice translation-rule 201
rule 1 /^\+.*/ //
rule 2 /^602.*/ //
rule 3 /^1512.*/ //
rule 4 /^00.*/ //
rule 5 /^0011.*/ //
rule 6 /^512/ /1&/
rule 7 /^1206.*/ /&/
rule 8 /^011\(31.*\)/ /\1/
voice translation-rule 202
rule 1 /^1\(5126026222\)$/ /\1/ type any subscriber plan any isdn
rule 2 /^1\(2065015555\)$/ /\1/ type any national plan any isdn
rule 3 /^1\(2065015151\)$/ /\1/ type any national plan any isdn
rule 4 /^1\(2065015111\)$/ /\1/ type any national plan any isdn
rule 5 /^31670357575$/ /&/ type any international plan any isdn
rule 6 /^31207037333$/ /&/ type any international plan any isdn
rule 7 /^31107047444$/ /&/ type any international plan any isdn
rule 8 /^911$/ /&/ type any unknown plan any unknown
rule 9 /^12065011.../ /&/ type any unknown plan any unknown
rule 10 /^31207033.../ /&/ type any unknown plan any unknown
rule 11 /^....$/ /&/ type any unknown plan any unknown
voice translation-rule 203
rule 1 /^512.*/ /&/ type any subscriber plan any isdn
rule 2 /^1/ // type any national plan any isdn
rule 3 /^00/ // type any international plan any isdn
voice translation-rule 301
rule 1 /^\+.*/ //
rule 2 /^20.*/ //
rule 3 /^0\([1-8].*\)/ /31\1/
rule 4 /^011/ //
rule 5 /^0031/ //
rule 6 /^703..../ /3120&/
rule 7 /^00\(1.*\)/ /\1/
voice translation-rule 302
rule 1 /^31207037333$/ /7037333/ type any subscriber plan any isdn
rule 2 /^7033\(...\)$/ /0207033\1/
rule 3 /^911$/ /112/ type any unknown plan any unknown
rule 4 /^31\(670357575\)$/ /0\1/ type any national plan any isdn
rule 5 /^31\(107047444\)$/ /0\1/ type any national plan any isdn
rule 6 /^12065015555$/ /&/ type any international plan any isdn
rule 7 /^12065015151$/ /&/ type any international plan any isdn
rule 8 /^12065015111$/ /&/ type any international plan any isdn
rule 9 /^15126026222$/ /&/ type any international plan any isdn
rule 10 /^12065011...$/ /&/ type any unknown plan any unknown
rule 11 /^15126022...$/ /&/ type any unknown plan any unknown
rule 12 /^....$/ /&/ type any unknown plan any unknown
voice translation-rule 303
rule 1 /^703.*/ /&/ type any subscriber plan any isdn
rule 2 /^010/ // type any national plan any isdn
rule 3 /^1/ // type any international plan any isdn
voice translation-rule 1000
rule 1 /.*\(1...$\)/ /206501\1/
rule 2 /.*\(2...$\)/ /512602\1/
rule 3 /.*\(45..$\)/ /020757\1/
voice translation-rule 1001
rule 1 /^1206...5...$/ /+&/
rule 2 /^1512...6...$/ /+&/
rule 3 /^31.0...7...$/ /+&/
voice translation-profile 1-HQ-Change_DNIS-Check_ANI
translate called 101
voice translation-profile 1-HQ-Proper_Types
translate calling 102
translate called 103
voice translation-profile 2-BR1-Change_DNIS-Check_ANI
translate called 201
voice translation-profile 2-BR1-Proper_Types
translate calling 202
translate called 203
voice translation-profile 3-BR2-Change_DNIS-Check_ANI
translate called 301
voice translation-profile 3-BR2-Proper_Types
translate calling 302
translate called 303
voice translation-profile SIP-NORMALIZE-DNIS-ANI
translate calling 1001
translate called 1000
voice-card 0
dspfarm
archive
log config
hidekeys
controller E1 0/0/0
clock source internal
pri-group timeslots 1-3,16
description == Voice Circuit to Branch2
controller T1 0/1/0
clock source internal
cablelength long 0db
pri-group timeslots 1-3,24
description == Voice Circuit to CorpHQ
controller T1 0/1/1
clock source internal
cablelength long 0db
pri-group timeslots 1-3,24
description == Voice Circuit to Branch1
interface Loopback0
ip address 177.1.254.254 255.255.255.255
interface Loopback10
ip address 177.1.254.250 255.255.255.255
interface Loopback11
ip address 177.1.254.251 255.255.255.255
interface FastEthernet0/0
description ==TO INTERNET==
ip address 192.168.1.150 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description === To HQ
ip address 177.1.19.1 255.255.255.0
duplex auto
speed auto
interface Serial0/0/0:15
description == PRI Circuit to R3-BR2
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn protocol-emulate network
isdn incoming-voice voice
isdn negotiate-bchan resend-setup
no isdn outgoing ie network-facility
isdn outgoing display-ie
no cdp enable
interface Serial0/1/0:23
description == PRI Circuit to R1-HQ
no ip address
encapsulation hdlc
isdn switch-type primary-5ess
isdn protocol-emulate network
isdn incoming-voice voice
isdn negotiate-bchan
isdn outgoing display-ie
no cdp enable
interface Serial0/1/1:23
description == PRI Circuit to R2-BR1
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn protocol-emulate network
isdn incoming-voice voice
isdn supp-service name calling
isdn negotiate-bchan resend-setup
isdn outgoing ie network-facility
no cdp enable
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.1.1
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
control-plane
voice-port 0/0/0:15
translation-profile incoming 3-BR2-Change_DNIS-Check_ANI
description == Voice PRI to Branch2
voice-port 0/1/0:23
translation-profile incoming 1-HQ-Change_DNIS-Check_ANI
description == Voice PRI to CorpHQ
voice-port 0/1/1:23
translation-profile incoming 2-BR1-Change_DNIS-Check_ANI
description == Voice PRI to Branch1
dial-peer voice 1 pots
description == All inbound calls from HQ BR1 BR2 into PSTN
incoming called-number .
direct-inward-dial
dial-peer voice 101 pots
description == Subscriber Calls from PSTN into CorpHQ
translation-profile outgoing 1-HQ-Proper_Types
preference 1
destination-pattern ^2065011...$
direct-inward-dial
port 0/1/0:23
forward-digits 10
dial-peer voice 102 pots
description == National Calls from PSTN into CorpHQ
translation-profile outgoing 1-HQ-Proper_Types
preference 1
destination-pattern ^12065011...$
direct-inward-dial
port 0/1/0:23
forward-digits 10
dial-peer voice 103 pots
description == International Calls into CorpHQ from PSTN Coming from NL Ph
translation-profile outgoing 1-HQ-Proper_Types
preference 1
destination-pattern ^0012065011...$
direct-inward-dial
port 0/1/0:23
forward-digits 10
dial-peer voice 104 pots
description == + Calls into CorpHQ from PSTN Coming from Mobiles
translation-profile outgoing 1-HQ-Proper_Types
preference 1
destination-pattern +12065011...$
direct-inward-dial
port 0/1/0:23
forward-digits 10
dial-peer voice 201 pots
description == Subscriber Calls from PSTN into Branch1
translation-profile outgoing 2-BR1-Proper_Types
preference 1
destination-pattern ^5126022...$
direct-inward-dial
port 0/1/1:23
forward-digits 10
dial-peer voice 202 pots
description == National Calls from PSTN into Branch1
translation-profile outgoing 2-BR1-Proper_Types
preference 1
destination-pattern ^15126022...$
direct-inward-dial
port 0/1/1:23
forward-digits 10
dial-peer voice 203 pots
description == International Calls into Branch1 from PSTN Coming from NL Ph
translation-profile outgoing 2-BR1-Proper_Types
preference 1
destination-pattern ^0015126022...$
direct-inward-dial
port 0/1/1:23
forward-digits 10
dial-peer voice 204 pots
description == + Calls into Branch1 from PSTN Coming from Mobiles
translation-profile outgoing 2-BR1-Proper_Types
preference 1
destination-pattern +15126022...$
direct-inward-dial
port 0/1/1:23
forward-digits 10
dial-peer voice 301 pots
description == Subscriber Calls from PSTN into Branch2
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern ^7033...$
direct-inward-dial
port 0/0/0:15
forward-digits 7
dial-peer voice 302 pots
description == National Calls from PSTN into Branch2
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern ^0207033...$
direct-inward-dial
port 0/0/0:15
forward-digits 10
dial-peer voice 303 pots
description == International Calls into Branch2 from PSTN Coming from US Ph
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern ^01131207033...$
direct-inward-dial
port 0/0/0:15
forward-digits 9
prefix 0
dial-peer voice 304 pots
description == International Calls into Branch2 from PSTN Coming from US Ph
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern ^31207033...$
direct-inward-dial
port 0/0/0:15
forward-digits 9
prefix 0
dial-peer voice 305 pots
description == + Calls into Branch2 from PSTN Coming from Mobiles
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern +31207033...$
direct-inward-dial
port 0/0/0:15
forward-digits 9
prefix 0
dial-peer voice 1000 voip
description == Calls into AT&T SIP ITSP for VC Week1 Lab1
rtp payload-type nse 99
rtp payload-type nte 100
voice-class sip localhost dns:sip1.att.com
session protocol sipv2
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 5000 voip
service aa
destination-pattern A5000
session target ipv4:177.1.254.254
incoming called-number A5000
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
num-exp 1888....... 911
num-exp 1900....... 911
num-exp 1976....... 911
num-exp 1777....... 911
num-exp 1444....... 911
num-exp 0800....... 911
num-exp 0900....... 911
sip-ua
telephony-service
no auto-reg-ephone
max-ephones 1
max-dn 10
ip source-address 177.1.254.254 port 2000
caller-id block code *67
system message You WILL PASS this Exam!
voicemail A5000
max-conferences 8 gain -6
call-forward pattern .T
dn-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Sep 01 2012 15:29:37
ephone-dn 1 dual-line
number 12065015111 secondary +12065015111
label Seattle, US +1 206 501 5111
description INE PSTN Phone
name Seattle US Phone
ephone-dn 2 dual-line
number 15126026222 secondary +15126026222
label Austin, US +1 512 602 6222
name Austin TX Phone
ephone-dn 3 dual-line
number 31207037333 secondary +31207037333
label Amsterdam, NL +31 20 703 73 33
name Amsterdam NL Phone
ephone-dn 4 dual-line
number 12065015555 secondary +12065015555
label Hurley Mobile +1 206 501 5555
name Hurley's Mobile
call-forward busy A5000
call-forward noan A5000 timeout 16
ephone-dn 5 dual-line
number 12065015151 secondary +12065015151
label Hurley's Home +1 206 501 5151
name Hurley's Home
call-forward busy A5000
call-forward noan A5000 timeout 12
ephone-dn 6 dual-line
number 31670357575 secondary +31670357575
label Sawyer's Mobile +31 6 70357575
name Sawyer's Mobile
call-forward busy A5000
call-forward noan A5000 timeout 16
ephone-dn 7 dual-line
number 911 secondary 112
label US/EU Emer/FreePhone/Prem
name Emergency Services
ephone-dn 8 dual-line
number 15126026262 secondary +15126026262
label BLinus Mobile +1 512 602 6262
name Benjamin Linus Mobile
call-forward busy A5000
call-forward noan A5000 timeout 16
ephone-dn 9 dual-line
number 31207037373 secondary +31207037373
label DHume Home +31 20 703 73 73
name Desmond Hume Home
call-forward busy A5000
call-forward noan A5000 timeout 16
ephone-dn 10 dual-line
number 31107047444 secondary +31107047444
label Rotterdam, NL +31 10 704 74 44
name Rotterdam NL Phone
ephone 1
device-security-mode none
mac-address A456.3040.0DAA
type 7975
button 1:1 2:2 3:3 4:10
button 5:6 6o7,8,5,4
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous level 0 limit 20
line aux 0
line vty 0 4
exec-timeout 0 0
privilege level 15
logging synchronous
no login
line vty 5 15
exec-timeout 0 0
privilege level 15
logging synchronous
no login
scheduler allocate 20000 1000
ntp source Loopback0
ntp master 10
ntp server 64.90.182.55
end
CorpHQ#sh run
Building configuration...
Current configuration : 6353 bytes
! No configuration change since last restart
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
hostname CorpHQ
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone PST -8
clock summer-time PDT recurring
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 177.1.11.1 177.1.11.14
ip dhcp excluded-address 177.1.11.21 177.1.11.254
ip dhcp excluded-address 177.2.11.1 177.2.11.14
ip dhcp excluded-address 177.2.11.21 177.2.11.254
ip dhcp pool CorpHQ-Phones
network 177.1.11.0 255.255.255.0
option 150 ip 177.1.10.10 177.1.10.20
default-router 177.1.11.1
dns-server 177.1.100.110
ip dhcp pool Branch1-Phones
network 177.2.11.0 255.255.255.0
option 150 ip 177.1.10.10 177.1.10.20
default-router 177.2.11.1
dns-server 177.1.100.110
no ip domain lookup
ip multicast-routing
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-ni
voice service voip
allow-connections h323 to h323
fax protocol cisco
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
no update-callerid
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
voice translation-rule 1
rule 1 // // type any subscriber plan any isdn
voice translation-rule 2
rule 1 // // type any national plan any isdn
voice translation-rule 3
rule 1 // // type any international plan any isdn
voice translation-rule 10
rule 1 /^[2-9].........$/ /9&/
rule 2 /^1[2-9].........$/ /9&/
rule 3 /^011/ /9&/
voice translation-profile MakeInternational
translate called 3
voice translation-profile MakeNational
translate called 2
voice translation-profile MakeSubscriber
translate called 1
voice translation-profile Prefix9_InFrom_CUCM
translate called 10
voice-card 0
dsp services dspfarm
archive
log config
hidekeys
controller T1 0/0/0
pri-group timeslots 1-3,24
description == Voice Circuit to PSTN
interface Loopback0
ip address 177.1.254.1 255.255.255.255
ip pim dense-mode
interface FastEthernet0/0
description == To CorpHQ-Switch
no ip address
duplex auto
speed auto
interface FastEthernet0/0.10
description == Server VLAN
encapsulation dot1Q 10
ip address 177.1.10.1 255.255.255.0
ip pim dense-mode
interface FastEthernet0/0.11
description == Voice VLAN
encapsulation dot1Q 11
ip address 177.1.11.1 255.255.255.0
ip helper-address 177.1.10.10
ip nbar protocol-discovery
ip pim dense-mode
interface FastEthernet0/0.12
description == Data VLAN
encapsulation dot1Q 12
ip address 177.1.12.1 255.255.255.0
interface FastEthernet0/0.13
description == PSTN PHONE VLAN
encapsulation dot1Q 13
ip address 177.1.13.1 255.255.255.0
interface FastEthernet0/1
description === To PSTN
ip address 177.1.19.254 255.255.255.0
duplex auto
speed auto
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
no cdp enable
interface Serial0/1/0
description == Frame-Relay Circuit to WAN
no ip address
encapsulation frame-relay
fair-queue 64 256 36
cdp enable
frame-relay lmi-type ansi
ip rsvp bandwidth
interface Serial0/1/0.1 point-to-point
description == FR To BR1
bandwidth 384
ip address 177.0.101.1 255.255.255.0
ip pim dense-mode
snmp trap link-status
frame-relay interface-dlci 101
ip rsvp bandwidth 136
interface Serial0/1/0.2 point-to-point
description == FR To BR2
ip address 177.0.201.1 255.255.255.0
snmp trap link-status
frame-relay interface-dlci 102
ip rsvp bandwidth 136
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 177.1.19.1
ip route 0.0.0.0 0.0.0.0 FastEthernet0/0.10
no ip http server
no ip http secure-server
control-plane
voice-port 0/0/0:23
voice-port 0/3/0
voice-port 0/3/1
ccm-manager music-on-hold
sccp local Loopback0
sccp ccm 177.1.10.10 identifier 1 priority 1 version 5.0.1
sccp ccm 177.1.10.20 identifier 2 priority 2 version 5.0.1
sccp ccm 177.1.254.3 identifier 3 priority 3 version 5.0.1
sccp
sccp ccm group 1
bind interface Loopback0
associate ccm 2 priority 1
associate ccm 1 priority 2
associate ccm 3 priority 3
associate profile 1 register CorpHQ-729-MTP
associate profile 2 register CorpHQ-711-MTP
associate profile 3 register CorpHQ-HW-Xcode
dspfarm profile 3 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
codec ilbc
maximum sessions 2
associate application SCCP
dspfarm profile 1 mtp
codec g729ar8
codec g729r8
rsvp
maximum sessions software 10
associate application SCCP
dspfarm profile 2 mtp
codec g711ulaw
rsvp
maximum sessions software 10
associate application SCCP
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
dial-peer voice 10 pots
translation-profile outgoing MakeSubscriber
destination-pattern 911
no digit-strip
port 0/0/0:23
dial-peer voice 11 pots
translation-profile outgoing MakeSubscriber
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
dial-peer voice 12 pots
translation-profile outgoing MakeNational
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 11
dial-peer voice 13 pots
translation-profile outgoing MakeInternational
destination-pattern 9011T
port 0/0/0:23
prefix 011
dial-peer voice 100 voip
description == Inbound/Outbound SIP PSTN GW From/To CUCM Pub
translation-profile incoming Prefix9_InFrom_CUCM
destination-pattern ^2065011...$
voice-class codec 1
session protocol sipv2
session target ipv4:177.1.10.10
incoming called-number .
ip qos dscp cs3 signaling
dial-peer hunt 1
sip-ua
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous level 0 limit 20
line aux 0
line vty 0 4
exec-timeout 0 0
privilege level 15
logging synchronous
no login
line vty 5 15
exec-timeout 0 0
privilege level 15
logging synchronous
no login
scheduler allocate 20000 1000
ntp source Loopback0
ntp master 2
ntp server 177.1.254.254
end
Branch1#sh run
Building configuration...
Current configuration : 3838 bytes
! Last configuration change at 01:19:02 CDT Thu Oct 10 2013
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
hostname Branch1
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone CST -6
clock summer-time CDT recurring
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
dot11 syslog
ip source-route
ip cef
ip multicast-routing
no ipv6 cef
ntp update-calendar
ntp server 177.1.254.1
multilink bundle-name authenticated
isdn switch-type primary-ni
voice-card 0
dsp services dspfarm
archive
log config
hidekeys
controller T1 0/0/0
pri-group timeslots 1-3,24 service mgcp
interface Loopback0
ip address 177.1.254.2 255.255.255.255
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.11
description == Voice VLAN
encapsulation dot1Q 11
ip address 177.2.11.1 255.255.255.0
ip helper-address 177.1.254.1
ip pim dense-mode
interface FastEthernet0/0.12
description == Data VLAN
encapsulation dot1Q 12
ip address 177.2.12.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn supp-service name calling
isdn bind-l3 ccm-manager
isdn outgoing ie facility
isdn outgoing display-ie
isdn outgoing ie redirecting-number
no cdp enable
interface Serial0/1/0
description == Frame-Relay Circuit to WAN
no ip address
encapsulation frame-relay
fair-queue 64 256 37
cdp enable
no frame-relay inverse-arp
frame-relay lmi-type ansi
ip rsvp bandwidth
interface Serial0/1/0.1 point-to-point
description == FR To HQ
ip address 177.0.101.2 255.255.255.0
ip pim dense-mode
snmp trap link-status
frame-relay interface-dlci 101
ip rsvp bandwidth 136
interface Serial0/1/1
no ip address
shutdown
clock rate 2000000
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
ip forward-protocol nd
no ip http server
no ip http secure-server
control-plane
voice-port 0/0/0:23
ccm-manager fallback-mgcp
ccm-manager redundant-host 177.1.10.10
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
mgcp
mgcp call-agent 177.1.10.20 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp fax t38 ecm
mgcp bind control source-interface Loopback0
mgcp bind media source-interface Loopback0
mgcp profile default
sccp local Loopback0
sccp ccm 177.1.254.3 identifier 3 priority 3 version 5.0.1
sccp ccm 177.1.10.10 identifier 1 priority 1 version 5.0.1
sccp ccm 177.1.10.20 identifier 2 priority 2 version 5.0.1
sccp
sccp ccm group 1
bind interface Loopback0
associate ccm 2 priority 1
associate ccm 1 priority 2
associate ccm 3 priority 3
associate profile 3 register Br1-HW-Xcode
associate profile 1 register Br1-729-MTP
associate profile 2 register Br1-711-MTP
dspfarm profile 3 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dspfarm profile 1 mtp
codec g729ar8
codec g729r8
rsvp
maximum sessions software 10
associate application SCCP
dspfarm profile 2 mtp
codec g711ulaw
rsvp
maximum sessions software 10
associate application SCCP
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous level 0 limit 20
line aux 0
line vty 0 4
exec-timeout 0 0
privilege level 15
logging synchronous
no login
line vty 5 15
exec-timeout 0 0
privilege level 15
logging synchronous
no login
scheduler allocate 20000 1000
end
Branch2#sh run
Building configuration...
Current configuration : 5789 bytes
! No configuration change since last restart
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
hostname Branch2
boot-start-marker
boot system flash:c2800nm-advipservicesk9-mz.124-24.T7.bin
boot system flash:
boot-end-marker
card type e1 0 0
logging message-counter syslog
no aaa new-model
clock timezone CEST 1
clock summer-time CEDT recurring
network-clock-participate wic 0
no network-clock-participate aim 0
dot11 syslog
ip source-route
ip cef
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-net5
voice service voip
no supplementary-service h225-notify cid-update
fax protocol cisco
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
voice class custom-cptone JOIN-TONE
dualtone conference
frequency 300 3600
cadence 150 100 500
voice class custom-cptone LEAVE-TONE
dualtone conference
frequency 300 3600
cadence 500 100 150
voice translation-rule 1
rule 1 /^7033...$/ /020&/
voice translation-rule 10
rule 1 /^0/ /0&/
voice translation-rule 200
rule 1 /^206501...$/ /1&/
voice translation-profile 7DigitDNIS-to-10Digit
translate called 1
voice translation-profile Prefix0_InFrom_CUCM
translate called 10
voice translation-profile Prefix1-toCorpHQ-ANI
translate calling 200
voice-card 0
dsp services dspfarm
archive
log config
hidekeys
controller E1 0/0/0
pri-group timeslots 1-3,16
description == Voice Circuit to PSTN
controller E1 0/0/1
interface Loopback0
ip address 177.1.254.3 255.255.255.255
h323-gateway voip bind srcaddr 177.1.254.3
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.11
encapsulation dot1Q 11
ip address 177.3.11.1 255.255.255.0
ip helper-address 177.1.10.10
interface FastEthernet0/0.12
encapsulation dot1Q 12
ip address 177.3.12.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bchan-number-order ascending
no cdp enable
interface Serial0/1/0
description == Frame-Relay Circuit to WAN
no ip address
encapsulation frame-relay
fair-queue 64 256 37
cdp enable
no frame-relay inverse-arp
frame-relay lmi-type ansi
ip rsvp bandwidth
interface Serial0/1/0.1 point-to-point
description == FR To HQ
ip address 177.0.201.2 255.255.255.0
snmp trap link-status
frame-relay interface-dlci 102
ip rsvp bandwidth 136
interface Serial0/1/1
no ip address
shutdown
clock rate 2000000
interface Service-Engine1/0
no ip address
shutdown
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
ip forward-protocol nd
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
control-plane
voice-port 0/0/0:15
translation-profile incoming 7DigitDNIS-to-10Digit
ccm-manager music-on-hold
sccp local Loopback0
sccp ccm 177.1.10.20 identifier 2 priority 2 version 5.0.1
sccp ccm 177.1.10.10 identifier 1 priority 1 version 5.0.1
sccp ccm 177.1.254.3 identifier 3 priority 3 version 5.0.1
sccp
sccp ccm group 1
bind interface Loopback0
associate ccm 2 priority 1
associate ccm 1 priority 2
associate ccm 3 priority 3
associate profile 4 register Br2-HW-Conf
associate profile 3 register Br2-HW-Xcode
associate profile 2 register Br2-711-MTP
associate profile 1 register Br2-729-MTP
dspfarm profile 3 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dspfarm profile 4 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 1
conference-join custom-cptone JOIN-TONE
conference-leave custom-cptone LEAVE-TONE
associate application SCCP
dspfarm profile 1 mtp
codec g729ar8
codec g729r8
rsvp
maximum sessions software 10
associate application SCCP
dspfarm profile 2 mtp
codec g711ulaw
rsvp
maximum sessions software 10
associate application SCCP
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
dial-peer voice 10 pots
destination-pattern 112
no digit-strip
port 0/0/0:15
dial-peer voice 11 pots
destination-pattern 00[1-9]T
port 0/0/0:15
prefix 0
dial-peer voice 12 pots
translation-profile outgoing Prefix1-toCorpHQ-ANI
destination-pattern 000T
port 0/0/0:15
prefix 00
dial-peer voice 100 voip
description == Inbound/Outbound H323 PSTN GW From/To GK and CUCM Pub
translation-profile incoming Prefix0_InFrom_CUCM
destination-pattern 0207033...$
voice-class codec 1
session target ipv4:177.1.10.10
incoming called-number .
ip qos dscp cs3 signaling
dial-peer voice 101 voip
description == Outbound H323 PSTN GW To CUCM Sub
destination-pattern 0207033...$
voice-class codec 1
session target ipv4:177.1.10.20
ip qos dscp cs3 signaling
dial-peer hunt 1
telephony-service
max-ephones 1
max-dn 1
ip source-address 177.1.254.3 port 2000
max-conferences 8 gain -6
moh test.au
multicast moh 239.2.1.1 port 16384 route 177.1.254.3 177.3.11.1
transfer-system full-consult
create cnf-files version-stamp 7960 Sep 13 2013 18:55:27
line con 0
exec-timeout 0 0
line aux 0
line 66
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
line vty 0 4
login
scheduler allocate 20000 1000
ntp source Loopback0
ntp update-calendar
ntp server 177.1.254.1
end -
Hello Everyone,
This is a post to Cisco for a bit of feedback regarding the BE3000. I've been using the device in the lab for the past few days and I wanted to make the follwoing points regarding the BE3K.
Just outline where I'm coming from I have a BE 3K with 8.6.4 installed with UK locale. I'm used to designing and deploying CUCM, CUC, CUPS, and integration with UCCX and UCCE, so I like to see alot of customisation in my CUCMs.
The redial key with outbound calls has a 6 second timer before the call connects. I have tested this with the Cisco 6945 which displays nothing but after 6 seconds the redial number displays and the call connects, and the Cisco 3905 which displays the last dialled number but again takes 6 seconds to connect.
Voicemail on the BE3K is very limited. It is a bit annoying to not be able to access Unity voicemail on the server. I would have liked to be able to see the users mailbox with the ability to import messages using Media Master (if the box is running Unity Connection) or set custom settings on the mailbox such as the behaviour after playing greetings. This could be most beneficial to add extra messages to an IVR where dummy phones could be inserted into the system to relay messages to voicemail for information recordings.
Fax with the VG224 and a SIP ITSP is very tempremental. Now this would be the same for any implementation of fax over SIP with a CUCM, it is never going to be easy, but it would be beneficial to be able to setup the VG224 using MGCP to see if this asssits the issue. Currently I the VG224 can be added as a SCCP device, which is great for analogue handsets, but it could be useful to have MGCP.
Updating from USB is very problematic. Installation of COP files hangs and times out requiring a reboot of the server. Luckily you can use SFTP with no issue.
The lack of customisation for SIP trunks without the SIP Trunk Connection Generic COP file. It is frustrating to see a wide variety of SIP configuration options in the documents and not be able to implement them without the COP file that unlocks advanced configuration. I understandthat Cisco are trying to make deploying the boxes easier for less technical users but teh absence of these options without installing the COP file which cannot be downloaded via the software page is unacceptable.
The boot time for the BE 3000 is unbelievably long compared to CUCM. I've watched the boot sequence and this takes a while but the activation of services once the CUCM has loaded the OS is very long. I imagine it's to do with the hardware of the BE 3K so it's not likely to get faster, but it is really slow.
The lack of options on the CLI. There is no ability to run the standard "show", "utils", and "set" commands on the CLI. The ability to issue the "utils service list paged" command would be beneficial.
This seems like I'm poking holes in the device but I actually believe it could be really good. There is a lot about the box that I like, it is great for unexperienced users, it's a breeze to deploy (I set my one up in less than a day with a customised dial plan), and it's competitively priced for the SMB. I'd like to see more attention to the Unity and IVR on the server, and the unlocking of the advanced configuration options.
It would be really good for Cisco to release a roadmap for the features due to be implemented on the BE3K, or perhaps if it exists someone could point me towards it.
Cheers,
James McMichaelJames & Adam,
Thanks for the feedback. It helps us greatly in creating the roadmap for the product.
Responding to some of your specific points:
1) As Adam mentioned, please clarify whether it is a PSTN call or an internal number. There may be some delay in getting the alert/progress from the PSTN network, if it is a pstn call. I tested on myself, and there is some delay, if redial is to a PSTN number.
3) Will look into this, Meanwhile the FAX solution is to use FXS on SPA8800.
4) Will see if there is an problem with uploading COP file using USB.
5) The Cisco supported configuration of SIP trunk is to connect to a CUBE. The configuration options which are exposed on SIP trunk page are configuration items which are required, if SIP trunk is connected to a CUBE. There is a smart design guide which provides details on what configuration is required on CUBE.
If you are not connecting SIP trunk to CUBE, then you would require advanced connection pack, if you want to expose more configurable options for SIP trunk settings.
6) We understand that the boot time is high. It should be reduced in upcoming releases.
2 & 7) These items are asking for advanced configuration options, which partners like you, who have worked with CUCM, CUP etc. are used to. This product is mainly targeted to SMB customers or partners who may not have that kind of expertise with advanced configuration options.
We understand that it is important to meet requirement from advanced partners also. We will take this feedback and see how we can expose these parameters and keeping the system simplified at the same time. I do not have any timeline on when it will be done, since there is lof ot work required to fix things like high reboot and upgrade time.
Thanks -
I setup InformaCast 8.3 Basic Paging with CallManager 9.1 using SIP 7965 phones. The 7965s are in a Recipient Group with a DialCast number setup for that Recipient Group;
When I dial the CTI Route Point I hear;
Welcome to the Singlewire Informacast System...ding..ding
Were sorry no devices could be activated. Your broadcast will not be completed.Amir,
The resolution I found was to go to
Admin || Dial Cast || Dialing Configurations || Recipient Groups
and add an recipient group previously created under Recipients || Recipient Groups
Restart the CTI route point in Call Manager that Informacast is using.
I also spanned the port on the phone and did a packet capture to see if IGMPv2 packets were being received properly before and after the fix was implemented. -
CUBE - New Deployment Issue - Not working DTMF Relay
Hello,
Scheme:
Cisco SCCP-based IP Phone > CUCM 9.1 w/ SIP Trunk > CUBE (28XX, 151-4.M7) > SIP ITSP
CUCM Active Call Proc. Node IP: 10.10.10.9
CUBE Inside Interface IP: 10.10.10.10
CUBE Outside Interface IP: 20.20.20.20
Cisco IP Phone: 10.10.10.8
ITSP SBC IP: 30.30.30.30
ITSP SIP domain: itsp.domain
Calling Pty: 9017654321 (translated in CUCM's route pattern which addresses CUBE)
Called Pty: 9011234567
While call was connected calling party dialed consequently 0,1,2,3,4 but far-end IVR does not react :(
Symptom:
While outbound call is connected calling party (IP Phone) dials digits which are not detected by any far-end PSTN (non-corporate) IVR at all.
Thoughts:
ITSP support only inband relay (RFC2833, Named Telephone Events or NTEs).
Using NTE provides a standard way to transport DTMF tones in RTP packets.
Thus rtp-nte is configured for both CUCM and ITSP dial-peers on CUBE.
While initial troubleshooting found that for the active call inbound CUBE's leg shows rtp-nte, but outbound inband-voice.
A have an assumption that ITSP doesn't give us 101=rtp-nte payload in 183 Response but I'm not sure.
m=audio 10318 RTP/AVP 8
b=AS:64
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:20
Questions:
1.How to make CUBE to successfully relay DTMF in according to ITSP requirement?
2. Why 'show call act/hist voice brief' doesn't show call id? All my attempts are identified as 2... )
It is hard to differentiate b/w call active/history records..Ayodeji,
Thanks for your feedback.
If you look through the already attached output 'show call act voice' of the file 'case-no-dtmf_-_cube-show-20140210-1.txt' you will find the following:
PeerId=101
CallOrigin=2
tx_DtmfRelay=rtp-nte
CallDuration=00:00:05 sec
PeerId=201
CallOrigin=1
tx_DtmfRelay=inband-voice
CallDuration=00:00:05 sec
2 : 230 18:40:38.048 MSK Tue Feb 10 2015.1 +1890 pid:101 Answer 79017654321 active
dur 00:00:04 tx:234/37440 rx:233/37280
IP 10.10.10.8:22688 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711alaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
2 : 231 18:40:38.068 MSK Tue Feb 10 2015.1 +1860 pid:201 Originate 79011234567 active
dur 00:00:04 tx:233/37280 rx:307/49120
IP 30.30.30.30:10318 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711alaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
This means that the target dial-peers (inbound and outbound are matched as designed).
dial-peer voice 101 voip
description -= inbound leg from CUCM to CUBE =-
session protocol sipv2
incoming called-number x
voice-class codec 1
voice-class sip bind control source-interface Loopback0
voice-class sip bind media source-interface Loopback0
dtmf-relay rtp-nte
no vad
dial-peer voice 201 voip
description -= outbound leg from CUBE to ITSP =-
translation-profile outgoing cdpn-delete-prefix-00XX7
max-conn 40
destination-pattern x
session protocol sipv2
session target dns:sbc.itsp.domain
voice-class codec 1
voice-class sip profiles 1
voice-class sip bind control source-interface Vlan100
voice-class sip bind media source-interface Vlan100
dtmf-relay rtp-nte
no vad
Now I have an argue with ITSP to make them send me NTE in their 183/200 response..
I've also:
1. Tried to disable dtmf-relay at all on dial-peers (trying inband-voice) but this doesn't work and also not recommended AFAIK.
2. Changed the value for SIP Trunk DTMF Signaling Method from 'No preference' to 'RFC 2833' w/ reset applied recommended by Suresh. No luck. -
hello,
I have a spa942 phone and does not register with CUCM 6.0.Any idea on how to go about it.thanks.You may need a Cisco SIP license. I know the SPA942 will work with Cisco's UC520 platform, but I thought on Callmanager you needed a SIP license in order to register a phone.
Cory
CiscoBuy.com -
Hi guys,
got a question about t38.
I have a setup with a SIP trunk from an ITSP, a IOS router 2801 as the voice gw and an XMedius Xpress T38 fax server also using SIP.
I have been trying to get this working for about 2 weeks now without any success. I get the call through, inbound and outbound, but when the t38 fax server send a SIP REINVITE with T38 the router forwards the T38 reinvite over the ITSP SIP networkto the legacy fax machine and after a few seconds i get a 488 Not Acceptable Media SIP message.
Has anyone got t38 working with a SIP ITSP trunk terminated at an IOS voice gateway?
i have attached debug output for ccsip messages and voip ccapi inout.Yes... a friend of mine works at the ITSP... they also have t38 fax servers running in their network for hosting and internal use...
-
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
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