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CSCO12271274
Beginner

call forwarding issue with the calling number ID

I have CUCM 11.5 with sip trunk to  CUBE as CUCM>CUBE>SIP service provider, I have two situations at call forwarding:

1- I configure call forwarding for extension 7799 to forward calls to cell phone, so when I call from ext 7766 to ext 7799 I received at my cell phone 7766 as I choose at Outbound Calls on SIP trunk Calling Party Selection Originator so I received 6687766 the original calling party, this is good.

2-but when outside calling number try to call 7799 it can't forward the calls.

3- so I changed the  Calling Party Selection from Originator to Last Redirect Number(External) after that I am able to receive the forwarding calls on my cell phone but the problem here the calling number is 6687799 not the original calling party.

so what is the proper solution or workaround to solve this issue?

 

6 REPLIES 6
Vaijanath Sonvane
VIP Advocate

Hi,

Please check this post. I think you need to use second or third method depending on your service provider settings:

https://community.cisco.com/t5/collaboration-voice-and-video/configure-and-troubleshoot-call-forward-to-the-pstn-using-sip/ta-p/3118287 

 

 

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Thanks, Vaijanath S.

thank you for reply @Vaijanath Sonvane, actually, I read this document before and I tried the second and third method, but unfortunately neither of both succeeded so let me explain to you what I have done :

phone A is calling number from outside  is 6687373

Phone B is called number from my DID range is 8317799

Phone C is a cell phone that receives forward calls number 0541407232

at second method

  request INVITE sip-header P-Asserted-Identity add "P-Asserted Identity:<sip:8317755@10.68.62.114>"

where 8317755 is one my valid DID range and 10.68.62.114 is IP of my CUBE interface that faces ITSP

when phone A called phone B and waiting till forwarding, I still can hear ringing without the appearance of any call on phone B or Phone C

 

at third method 

request INVITE sip-header Diversion modify "<sip:(.*)@(.*)>" "<sip:8317756@10.68.62.114>"

when phone A called phone B and waiting till forwarding, I hear busy tone instead of forwarding the call

 

I take logs at the second and third methods you can find them in the attached files.

I want to make sure that the issue is not related to my configuration before going to ask my ITSP

 

Hi,

I looked at the logs and I do not see complete call signaling. The call signaling below doesn't seem to be right for third method.

Capture.JPG

 

Can you please post full configuration of your CUBE router? Also, please post the logs from CUCM and CUBE for same call for third method. 

 

 

Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.

Hi

kindly find the logs from CUCM and CUBE during the call and the configuration file for CUBE.

Hi,

I looked at the logs and the INVITE that is being sent from your CUBE router to service provider doesn't include DIVERSION header.

++++++++++++++++ Sent INVITE to Service Provider ++++++++++++++++

Sent: INVITE sip:0541407232@10.205.20.50:5060 SIP/2.0 Via: SIP/2.0/UDP 10.68.62.114:5060;branch=z9hG4bK39511FB8 Remote-Party-ID: <sip:6687373@10.68.62.114>;party=calling;screen=yes;privacy=off From: <sip:6687373@10.205.20.50>;tag=ABDF35F9-9F9 To: <sip:0541407232@10.205.20.50> Date: Tue, 01 Oct 2019 13:54:23 GMT Call-ID: CF05A706-E38911E9-AC5188D4-4E2674BC@10.68.62.114 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 4159592192-0000065536-0000006057-0253912074 User-Agent: Cisco-SIPGateway/IOS-15.4.3.S4 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1569938063 Contact: <sip:6687373@10.68.62.114:5060> Expires: 180 Allow-Events: telephone-event Max-Forwards: 65 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 247 v=0 o=CiscoSystemsSIP-GW-UserAgent 7261 5582 IN IP4 10.68.62.114 s=SIP Call c=IN IP4 10.68.62.114 t=0 0 m=audio 27586 RTP/AVP 0 101 c=IN IP4 10.68.62.114 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 ++++++++++++++++ Received TRYING from Service Provider ++++++++++++++++
007666: Oct 1 16:54:23.503: //1183833/F7EE57000000/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.68.62.114:5060;branch=z9hG4bK39511FB8 Call-ID: CF05A706-E38911E9-AC5188D4-4E2674BC@10.68.62.114 From: <sip:6687373@10.205.20.50>;tag=ABDF35F9-9F9 To: <sip:0541407232@10.205.20.50> CSeq: 101 INVITE Content-Length: 0 ++++++++++++++++ Received 484 ADDRESS INCOMPLETE from Service Provider ++++++++++++++++
007667: Oct 1 16:54:23.562: //1183833/F7EE57000000/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 10.68.62.114:5060;branch=z9hG4bK39511FB8 Record-Route: <sip:10.205.20.50:5060;transport=udp;lr> Call-ID: CF05A706-E38911E9-AC5188D4-4E2674BC@10.68.62.114 From: <sip:6687373@10.205.20.50>;tag=ABDF35F9-9F9 To: <sip:0541407232@10.205.20.50>;tag=sbc0806sbcthLNQHaQBLtqQ CSeq: 101 INVITE Reason: Q.850; cause=28; text="address incomplete" Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR" Content-Length: 0 007668: Oct 1 16:54:23.565: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

It seems that the dial-peer configured with sip profile is not matching for these calls. Use debug voip ccapi inout command to see which dial-peers are matching and correct the configuration.

 

 

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Thanks, Vaijanath S.

Hi,

I forgot to mention in my earlier post that you need to enable Diversion Header on CUCM SIP Trunk in order for CUBE Diversion Header configuration to work. In CUCM, go to Device > Trunk and find the SIP trunk that you configured for outgoing calls. In the Outbound Calls pane, ensure that the Redirecting Diversion Header Delivery - Outbound check box gets checked.

 

 

Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.
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