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Skype SIP Trunk Outgoing calls fail and no access to CUE

Hi,

Hi,
I have a UC540 configured with a Skype sip trunk. The trunk is registered with Skype. I'm unable to make outgoing calls. I configured the trunk using the latest version of CCA and followed and updated PDF doc for the Skype config. Incomning calls are working to a Skype online number. Im unable to make outgoing calls, and the the message button on the IP phones 7960's and CIPC running SCCP has stopped working. I ran debugs the output is below. First debug is an outgoing call second debug is an attempt to access voice mail when hitting the message button, this equates to connecting to cue at extension 399. Any advise would be appreciated. I have also attached a config comparison of the original working config and the config after the Skype CCA config ws applied.
Thanks
Andy

Below here is an outgoing call debug

home-uc540#
000845: Nov 29 23:03:39.201: //82/B922D683808A/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x87F5CA00
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 99051000122528
Called Number            : 3038100583
Source IP Address (Sig  ): 24.8.157.192
Destn SIP Req Addr:Port  : 63.209.144.201:5060
Destn SIP Resp Addr:Port : 63.209.144.201:5060
Destination Name         : sip.skype.com

home-uc540#
000846: Nov 29 23:03:39.201: //82/B922D683808A/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 24.8.157.192
Source IP Port    (Media): 18328
Destn  IP Address (Media):  -
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

000847: Nov 29 23:03:39.201: //82/B922D683808A/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 1
Disconnect Cause (SIP)   : 404

Below is the output when I push the message button and the phone dials 399 which is a sip dial-peer.

home-uc540#debug ccsip call


000837: Nov 29 23:00:03.171: //78/3BE113AF8084/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x87F5CA00
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 202
Called Number            : 399
Source IP Address (Sig  ): 10.1.10.2
Destn SIP Req Addr:Port  : 10.1.10.1:5060
Destn SIP Resp Addr:Port : 10.1.10.1:5060
Destination Name         : 10.1.10.1

home-uc540#
000838: Nov 29 23:00:03.171: //78/3BE113AF8084/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 10.1.10.2
Source IP Port    (Media): 18362
Destn  IP Address (Media):  -
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

000839: Nov 29 23:00:03.171: //78/3BE113AF8084/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 41
Disconnect Cause (SIP)   : 500

1 REPLY
New Member

Re: Skype SIP Trunk Outgoing calls fail and no access to CUE

Hi,

Well I figured out why the outgoing calls were failing. It turns out that Skype will not except a 10 digit number there needed to be a 1 in front so 11 digit. I was trying call a local 10 digit number 303-xxx-xxxx and the call was failing, in my local area 720-xxx-xxxx is also a local 10 digit area code. Any long distance numbers where I inserted a 1 were working, for example 1-408-xxx-xxxx. The solution was to edit the outgoing translational rule to add a 1 before the two local area codes of 303 and 720. This means that users can still dial the 10 digit local number without having to dial a 1.

The translational rule I used was,

voice translation-rule 1112
rule 1 /^9303/ /1303/
rule 2 /^9720/ /1720/
rule 3 /^9/ //

The above rule adds a 1  in front of the 9303 and removes the 9, same for 9720 it adds a 1 in front of the 9720.

The voice mail problem still exists I will continue to troubleshoot that.

Andy

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