Re: Call forwarding all not working

bozhidarpetkov1 · ‎05-13-2019

Hi,

apologies in advance, i'm not that familiar with voice technologies.

I have a problem where when an extension is setup with the call-forwarding all command to a mobile number and is rang from outside the company the ringing party hears an engaged tone. However, at the same time if a company member calls that extension from another extension the call is transferred to the mobile number with no problems.

The way the setup works is that user phone's are registered on a CME router running version 12.4(13r)T which then routes the calls to another CME router running version 15.0(1)M6 which terminates the SIP trunk from our provider.

I'm told forwarding worked just fine a few days ago. No configuration changes have been made and i have confirmed that from the logs.

I have attached both configs which have been trimmed down from unnecessary stuff. I did try to run some debugs but I'm uncertain which are best in this way so the output I got didn't give me much information.

Bobby

davidpatton · ‎05-13-2019

Can you provide the call setup details via "debug ccsip messages" from the CME / CUBE connected to the carrier? Most likely an issue in the INVITE with properly formatted calling party information.

bozhidarpetkov1 · ‎05-14-2019

I have run the debug as suggested and attached it to my reply.

HARIS_HUSSAIN · ‎05-14-2019

The Call is Being Forwarded to the CUBE Router, but we get 404 Not Found reply.

IT is sendign call as 07766512582 Make sure this is correct format, Usually we would add 9 for external calls. If that is the case then make changes in translation pattern to add 9. Also capture debug ccsip on the CUBE Router.

May 14 07:28:28.362: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:07766512582@88.215.61.194:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.6.220:5060;branch=z9hG4bKF2F01C84
Remote-Party-ID: <sip:+35989701784485202142@10.11.6.220>;party=calling;screen=yes;privacy=off
From: <sip:+35989701784485202142@10.11.6.220>;tag=8ED83240-133D
To: <sip:07766512582@88.215.61.194>
Date: Tue, 14 May 2019 07:28:28 GMT
Call-ID: B3A9B5E7-755011E9-BD379BB1-184774B7@10.11.6.220
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 5400
Cisco-Guid: 3013563918-1968181737-3174144945-0407336119
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1557818908
Contact: <sip:+35989701784485202142@10.11.6.220:5060>
Call-Info: <sip:10.11.6.220:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Diversion: <sip:224@10.11.6.220>;privacy=off;reason=unconditional;counter=1;screen=no
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires: 5400;refresher=uac
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 2710 1810 IN IP4 10.11.6.220
s=SIP Call
c=IN IP4 10.11.6.220
t=0 0
m=audio 19438 RTP/AVP 8 101
c=IN IP4 10.11.6.220
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

*May 14 07:28:28.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.11.6.220:5060;branch=z9hG4bKF2EF7BF
To: <sip:07766512582@88.215.61.194>;tag=3766807948-60572035
From: <sip:+35989701784485202142@10.11.6.220>;tag=8ED83204-15E2
Call-ID: B3A08E5E-755011E9-BD359BB1-184774B7@10.11.6.220
CSeq: 101 INVITE
Allow: UPDATE,PRACK,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Contact: <sip:07766512582@88.215.61.194:5060>
Content-Length: 0

bozhidarpetkov1 · ‎05-14-2019

The config under the extension is:

call-forward all 907766512582

I have attached another debug from the other router.

Bobby

davidpatton · ‎05-14-2019

For future posts please provide a clear call flow and some way of following the original configuration to call flow. It was redacted and incomplete so I had to piece together the call flow and configuration.

Basically, it appears the call is failing due to an invalid diversion header sent by CME1's 302 Moved message to CME2 that is sent to the Carrier. To fix this it should be as simple as an update to CME1:

voice service voip
no supplementary-service sip moved-temporarily

I've cobbled this together but to help, you'll have to provide more information or open a TAC case and share the detailed information with them.

Call flow:

CME1 > CME2 > 224 CFALL to 907766512582 > CME2 Dial-peer 1 destination-pattern 90[1-9]......... > Carrier

Carrier: 88.215.61.194

CME2 with CFALL from 224 and CUBE to PSTN: 10.11.6.220

CME1: Source IP 10.11.3.220 / Ethernet or LB on CME 10.11.6.221

1. You are getting an error from your carrier about the session timer being too small. You need to find out what they are expecting and fix that issue.

*May 14 07:28:28.226: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 422 Session Timer too small
Via: SIP/2.0/UDP 88.215.61.194:5060;branch=z9hG4bKcab49fa612f642ccd351349cd7ef22f0
From: <sip:+359897202142@88.215.61.194;user=phone>;tag=3766807948-1944865151
To: <sip:01784485224@88.215.61.194;user=phone>;tag=8ED831BC-258D
Date: Tue, 14 May 2019 07:28:28 GMT
Call-ID: 15882569-3766807948-840629610@MSX22.gammatelecom.com
CSeq: 1 INVITE
Allow-Events: telephone-event
Min-SE: 5400
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

2. CME1 sends CME2 an Invite for the new call leg (redirected) and CME2 sends an update SIP/2.0 302 Moved Temporarily, with Diversion header: Diversion: <sip:224@10.11.6.221>;reason=unconditional;counter=1

*May 14 07:28:28.286: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 302 Moved Temporarily
Date: Tue, 14 May 2019 07:29:57 GMT
From: <sip:+359897202142@10.11.6.220>;tag=8ED831DC-1423
Allow-Events: telephone-event
Diversion: <sip:224@10.11.6.221>;reason=unconditional;counter=1
Timestamp: 1557818908
Content-Length: 0
To: <sip:224@10.11.6.221>;tag=8ED6CFC4-1C2
Contact: <sip:907766512582@10.11.6.220>
Call-ID: B39A7436-755011E9-BD309BB1-184774B7@10.11.6.220
Via: SIP/2.0/UDP 10.11.6.220:5060;branch=z9hG4bKF2EE9BF
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE

3. Your invite gets updated with an invalid Diversion: Diversion: <sip:224@10.11.6.220>;privacy=off;reason=unconditional;counter=1;screen=no

*May 14 07:28:28.302: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:07766512582@88.215.61.194:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.6.220:5060;branch=z9hG4bKF2EF7BF
Remote-Party-ID: <sip:+35989701784485202142@10.11.6.220>;party=calling;screen=yes;privacy=off
From: <sip:+35989701784485202142@10.11.6.220>;tag=8ED83204-15E2
To: <sip:07766512582@88.215.61.194>
Date: Tue, 14 May 2019 07:28:28 GMT
Call-ID: B3A08E5E-755011E9-BD359BB1-184774B7@10.11.6.220
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 5400
Cisco-Guid: 3013563918-1968181737-3174144945-0407336119
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1557818908
Contact: <sip:+35989701784485202142@10.11.6.220:5060>
Call-Info: <sip:10.11.6.220:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Diversion: <sip:224@10.11.6.220>;privacy=off;reason=unconditional;counter=1;screen=no
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires: 5400;refresher=uac
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 243
v=0
o=CiscoSystemsSIP-GW-UserAgent 4736 557 IN IP4 10.11.6.220
s=SIP Call
c=IN IP4 10.11.6.220
t=0 0
m=audio 18588 RTP/AVP 8 101
c=IN IP4 10.11.6.220
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

I always recommend getting comfortable with TranslatorX. It will help you visually determine call flow. Gathering both logs simultaneously (time for both needs to be spot on for this to work, NTP) means you can see all call legs in one pane of glass.

bozhidarpetkov1 · ‎05-15-2019

Apologies for not making this clearer but I was unsure what information is required to troubleshoot this issue. As for the command you suggested - this already exists in the original configuration so that's not the problem. The session timer shouldn't be a reason for the call forwarding to fail as all other normal calls are going through fine.

I am unsure about the Diversion error and what the cause for this would be. Do you have any other suggestions? Thanks.

davidpatton · ‎05-15-2019

If I am not mistaken, as per your configuration "CME config.txt", no supplementary-service sip moved-temporarily, is not there. If it was then we would have a different problem, why is it sending the message. The originating CME (CME2), 10.11.3.220 / 10.11.6.221 is the issue in this call flow and the one you need to add that command to as it originated the 302 message as per your debugs. The remaining troubleshooting would come after it is in fact enabled. Please reference the image posted previously as it shows the call flow and IP addresses. It is enabled on CME1 (CME to the PSTN) but that's not the issue, it's the originator CME2 that is the issue.

bozhidarpetkov1 · ‎05-15-2019

Sorry, i thought you meant CME01. I have now added the command to CME02 as requested and have run the debugs again. Forwarding still fails as expected. Let me know if want me to run any other debugs.

bozhidarpetkov1 · ‎05-24-2019

Hi,

i still haven't managed to get this working. Do you have any other suggestions?

Thanks,

Bobby