Call Drop on Consult Transfer

[mmaamm238]

Hi,

I have a problem with consult call transfer. I have SIP trunk with ITSP. When I want to transfer an external call from 7965 to 7945 it drops but from 7945 to 7965 is successful. Both phones can call to each other and outside and blind transfer works without problem. My CUCM version is 11.5.

How can I troubleshoot it? Is it signaling or codec problem?

[Carlo Poggiarelli]

Hi,

Can you please activate a debug ccisip messages on your VG and transfer a call from 7945 to 7965  and then from 7965 to 7945?

Please post the output here.

Thanks a lot

Regards

Carlo

[mmaamm238]

It is last logs when calling from 7965 to 7945 and trying transfer:

.Sep 23 08:56:12: //557234/293A4611BCA7/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP dddddd:5060;branch=z9hG4bK48506afda312
From: <sip:dddddd>;tag=838177268
To: <sip:eeeeee>;tag=660EE6EC-147C
Date: Sat, 23 Sep 2023 04:26:12 GMT
Call-ID: 52534f80-50e168e4-484eb-2e5aa8c0@dddddd
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 464

v=0
o=CiscoSystemsSIP-GW-UserAgent 2591 9692 IN IP4 eeeeee
s=SIP Call
c=IN IP4 eeeeee
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 eeeeee
m=image 0 udptl t38
c=IN IP4 eeeeee
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy

.Sep 23 08:56:13: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
UPDATE sip:aaaaaa@eeeeee:5060 SIP/2.0
Via: SIP/2.0/UDP cccccc:5060;branch=z9hG4bK16cbb48148b81
From: <sip:710@cccccc>;tag=126059~5065a019-d8b8-4620-bc99-b88f9d4927d9-31698473
To: "aaaaaa" <sip:aaaaaa@eeeeee>;tag=660DC5B4-D75
Date: Sat, 23 Sep 2023 04:25:05 GMT
Call-ID: FD180EC1-58FF11EE-BC98CCC4-5A51A9C8@eeeeee
User-Agent: Cisco-CUCM11.5
Max-Forwards: 70
Supported: timer,resource-priority,replaces
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 UPDATE
Session-ID: 8a5bc1f73cbf467d865862ba31698481;remote=a17754dea08e4a7732d6a933ab126059
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=VIDEO_UNSPECIFIED
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
Min-SE: 1800
P-Asserted-Identity: "Person3" <sip:706@cccccc>
Remote-Party-ID: "Person3" <sip:706@cccccc>;party=calling;screen=yes;privacy=off
Contact: <sip:710@cccccc:5060>
Content-Length: 0

.Sep 23 08:56:13: //557222/FD150161BC92/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:aaaaaa@ffffff:5060 SIP/2.0
Via: SIP/2.0/UDP gggggg:5060;branch=z9hG4bK5A9C911B1
Remote-Party-ID: "Person2" <sip:710@gggggg>;party=calling;screen=yes;privacy=off
From: <sip:bbbbbb@gggggg>;tag=660DC5C8-1A5A
To: "aaaaaa" <sip:aaaaaa@ffffff>;tag=as0c8871a4
Date: Sat, 23 Sep 2023 04:26:13 GMT
Call-ID: 1ee0fbe570cd2d670df4042b38d33ef2@ffffff:5060
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 4246012257-1493111278-3163737284-1515301320
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1695443173
Contact: <sip:bbbbbb@gggggg:5060>
Expires: 180
Allow-Events: kpml, telephone-event
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 4186 2596 IN IP4 gggggg
s=SIP Call
c=IN IP4 gggggg
t=0 0
m=audio 17358 RTP/AVP 0 101
c=IN IP4 gggggg
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

.Sep 23 08:56:13: //557223/FD150161BC92/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP cccccc:5060;branch=z9hG4bK16cbb48148b81
From: <sip:710@cccccc>;tag=126059~5065a019-d8b8-4620-bc99-b88f9d4927d9-31698473
To: "aaaaaa" <sip:aaaaaa@eeeeee>;tag=660DC5B4-D75
Date: Sat, 23 Sep 2023 04:26:13 GMT
Call-ID: FD180EC1-58FF11EE-BC98CCC4-5A51A9C8@eeeeee
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 UPDATE
Allow-Events: kpml, telephone-event
Contact: <sip:aaaaaa@eeeeee:5060>
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Length: 0

.Sep 23 08:56:13: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:aaaaaa@eeeeee:5060 SIP/2.0
Via: SIP/2.0/UDP cccccc:5060;branch=z9hG4bK16cbc2622edb8
From: <sip:710@cccccc>;tag=126059~5065a019-d8b8-4620-bc99-b88f9d4927d9-31698473
To: "aaaaaa" <sip:aaaaaa@eeeeee>;tag=660DC5B4-D75
Date: Sat, 23 Sep 2023 04:25:05 GMT
Call-ID: FD180EC1-58FF11EE-BC98CCC4-5A51A9C8@eeeeee
User-Agent: Cisco-CUCM11.5
Max-Forwards: 70
CSeq: 103 BYE
Reason: Q.850;cause=47
Session-ID: 8a5bc1f73cbf467d865862ba31698481;remote=a17754dea08e4a7732d6a933ab126059
Content-Length: 0

.Sep 23 08:56:13: //557222/FD150161BC92/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP gggggg:5060;branch=z9hG4bK5A9C911B1;received=gggggg
From: <sip:bbbbbb@gggggg>;tag=660DC5C8-1A5A
To: "aaaaaa" <sip:aaaaaa@ffffff>;tag=as0c8871a4
Call-ID: 1ee0fbe570cd2d670df4042b38d33ef2@ffffff:5060
CSeq: 101 INVITE
Server: SIP Server 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:aaaaaa@ffffff:5060>
Content-Length: 0

.Sep 23 08:56:13: //557222/FD150161BC92/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:aaaaaa@ffffff:5060 SIP/2.0
Via: SIP/2.0/UDP gggggg:5060;branch=z9hG4bK5A9CA65
From: <sip:bbbbbb@gggggg>;tag=660DC5C8-1A5A
To: "aaaaaa" <sip:aaaaaa@ffffff>;tag=as0c8871a4
Date: Sat, 23 Sep 2023 04:26:13 GMT
Call-ID: 1ee0fbe570cd2d670df4042b38d33ef2@ffffff:5060
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1695443173
CSeq: 102 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=3426,OS=548160,PR=1330,OR=212800,PL=0,JI=0,LA=0,DU=68
Content-Length: 0

.Sep 23 08:56:13: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP cccccc:5060;branch=z9hG4bK16cbc2622edb8
From: <sip:710@cccccc>;tag=126059~5065a019-d8b8-4620-bc99-b88f9d4927d9-31698473
To: "aaaaaa" <sip:aaaaaa@eeeeee>;tag=660DC5B4-D75
Date: Sat, 23 Sep 2023 04:26:13 GMT
Call-ID: FD180EC1-58FF11EE-BC98CCC4-5A51A9C8@eeeeee
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=1330,OS=212800,PR=3426,OR=548160,PL=0,JI=0,LA=0,DU=68
Content-Length: 0

.Sep 23 08:56:13: //557222/FD150161BC92/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP gggggg:5060;branch=z9hG4bK5A9C911B1;received=gggggg
From: <sip:bbbbbb@gggggg>;tag=660DC5C8-1A5A
To: "aaaaaa" <sip:aaaaaa@ffffff>;tag=as0c8871a4
Call-ID: 1ee0fbe570cd2d670df4042b38d33ef2@ffffff:5060
CSeq: 101 INVITE
Server: SIP Server 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:aaaaaa@ffffff:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 231

v=0
o=sip 1436636385 1436636386 IN IP4 ffffff
s=SIP Server 1.0
c=IN IP4 ffffff
t=0 0
m=audio 47728 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

.Sep 23 08:56:13: //557222/FD150161BC92/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP gggggg:5060;branch=z9hG4bK5A9C911B1;received=gggggg
From: <sip:bbbbbb@gggggg>;tag=660DC5C8-1A5A
To: "aaaaaa" <sip:aaaaaa@ffffff>;tag=as0c8871a4
Call-ID: 1ee0fbe570cd2d670df4042b38d33ef2@ffffff:5060
CSeq: 101 INVITE
Server: SIP Server 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

.Sep 23 08:56:13: //557222/FD150161BC92/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP gggggg:5060;branch=z9hG4bK5A9CA65;received=gggggg
From: <sip:bbbbbb@gggggg>;tag=660DC5C8-1A5A
To: "aaaaaa" <sip:aaaaaa@ffffff>;tag=as0c8871a4
Call-ID: 1ee0fbe570cd2d670df4042b38d33ef2@ffffff:5060
CSeq: 102 BYE
Server: SIP Server 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

.Sep 23 08:56:13: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:aaaaaa@ffffff:5060 SIP/2.0
Via: SIP/2.0/UDP gggggg:5060;branch=z9hG4bK5A9C911B1
From: <sip:bbbbbb@gggggg>;tag=660DC5C8-1A5A
To: "aaaaaa" <sip:aaaaaa@ffffff>;tag=as0c8871a4
Date: Sat, 23 Sep 2023 04:26:13 GMT
Call-ID: 1ee0fbe570cd2d670df4042b38d33ef2@ffffff:5060
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0

[Roger Kallberg]

Please put the output into a text file that you attach to the post. It makes it much easier to work with your output.

[mmaamm238]

I attached last logs when calling from 7965 to 7945 and trying transfer.

I think this is important:

CSeq: 103 BYE
Reason: Q.850;cause=47

[Carlo Poggiarelli]

Hi,

Sure it is

On transfer message, the CUCM is sending a By with codec mismatch reason code.

You need to be sure that both Inboud and Outbound call legs are matching the correct DP and the right codec or codec class is configured.

Keep in mind that , if none of configured DP is matched, the VG/Cube uses the default DP (0) that uses the G729 codec

Can you please post your Dial-peer config here so we can check?

Thanks a lot

Regaeds

Carlo

[mmaamm238]

Here it is attached

[Roger Kallberg]

I assume that you have dual interfaces in use in your SBC, one for your ITSP and another for your internal network. If so you should have two pairs of dial peers, an inbound and another outbound used for the call leg on your internal network and another pair, one inbound and one outbound, dial peers for the ITSP. On these you should define bind statements for signaling and media.

Also you have not turned on the Cube functionality in your SBC. Without that it will not act as an SBC. From your output you have effectively turned off the security in your SBC as you allow all communication with 0.0.0.0 0.0.0.0 line in trusted networks. I strongly recommend you to remove that at the soonest.

[mmaamm238]

How can I turn on cube functionality?

[Roger Kallberg]

To enable Cube feature you do this, under "voice service voip" add "mode border-element". After that you need to reload the router to activate the functionality.

[mmaamm238]

There is no such command: mode border-element.

My router is 2821. What should I do?

[myonedbarayn]

---

@mmaamm238 wrote:

Hi,

I have a problem with consult call transfer. I have SIP trunk with ITSP. When I want to transfer an external call from 7965 to 7945 it drops but from 7945 to 7965 is successful. Both phones can call to each other and outside and blind transfer works without problem. My CUCM version is 11.5.

How can I troubleshoot it? Is it signaling or codec problem?

---

The issue you are experiencing with consult call transfer from 7965 to 7945 is most likely due to a problem with the signaling between your Cisco Unified Communications Manager (CUCM) and your ITSP.

Here are some things you can check:

• Make sure that the signaling path between your CUCM and your ITSP is configured correctly. This includes checking the following:
  • The IP addresses and ports that are being used for signaling
  • The codecs that are supported by both CUCM and your ITSP
  • The call routing rules that are in place
• You can also try to capture a packet trace of a consult call transfer from 7965 to 7945. This will help you to identify the exact point where the call is dropping.

[mmaamm238]

Thank you

The odd thing is consult transfer works from 7945 to 7965 and blind transfer works in both ways.

[Carlo Poggiarelli]

Hi,

Thanks for the dialpeer output.

Now I ask you to activate a debug voip dialpeer inout and make a consult transfer from 7945 to 7965 and then the opposite.

Thanks 

Carlo

[mmaamm238]

Here it is attached: