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calls CFA to voicemail fails from SIP Trunk

Hello

i got the next issue if you could help me would be great

We have a trunk SIP from CM 8.5 to Unity Connection 8.5 and its work fine, and also we have a SIP trunk to an ITSP that also works fine, the problem is when a user forward the phone to VoiceMail and someone calls from the PSTN to this user the call fail, i reviewed some logs and this is what i had.

Where 10.101.105.2 is the inside IP of CUBE, 10.101.105.6 is CM ip address, 172.31.37.166 is the ip address of the remote ITSP peer and the 172.22.17.227 is the Trunk SIP ip address. we have noted that is not negociating the codec but the voice class codec is present on all dial-peers

The Call Setup Information is:

Call Control Block (CCB) : 0x2BFD1C60

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 973796965

Called Number            : 25785496

Source IP Address (Sig  ): 10.101.105.2

Destn SIP Req Addr:Port  : 10.101.105.6:5060

Destn SIP Resp Addr:Port : 10.101.105.6:5060

Destination Name         : 10.101.105.6

Apr 20 16:53:34.975: //2160/33C3CE438E7D/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): 10.101.105.2

Source IP Port    (Media): 18772

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

Orig Destn IP Address:Port (Media): [ - ]:0

Apr 20 16:53:34.975: //2160/33C3CE438E7D/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 63

Disconnect Cause (SIP)   : 503

Apr 20 16:53:35.071: //2161/33C3CE438E7D/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x2BFB00C0

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 973796965

Called Number            : 25785496

Source IP Address (Sig  ): 10.101.105.2

Destn SIP Req Addr:Port  : 10.101.105.5:5060

Destn SIP Resp Addr:Port : 10.101.105.5:5060

Destination Name         : 10.101.105.5

Apr 20 16:53:35.071: //2161/33C3CE438E7D/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): 10.101.105.2

Source IP Port    (Media): 22182

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

Orig Destn IP Address:Port (Media): [ - ]:0

Apr 20 16:53:35.071: //2161/33C3CE438E7D/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 63

Disconnect Cause (SIP)   : 503

Apr 20 16:53:35.083: //2162/33C3CE438E7D/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x2BFF3800

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 973796965

Called Number            : 25785496

Source IP Address (Sig  ): 172.31.37.166

Destn SIP Req Addr:Port  : 172.22.17.227:5060

Destn SIP Resp Addr:Port : 172.22.17.227:5060

Destination Name         : 172.22.17.227

Apr 20 16:53:35.083: //2162/33C3CE438E7D/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): 172.31.37.166

Source IP Port    (Media): 31004

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

Orig Destn IP Address:Port (Media): [ - ]:0

Apr 20 16:53:35.083: //2162/33C3CE438E7D/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 18

Disconnect Cause (SIP)   : 480

Apr 20 16:53:35.099: //2159/33C3CE438E7D/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x2BFAA6D0

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 973796965

Called Number            : 25785496

Source IP Address (Sig  ): 172.31.37.166

Destn SIP Req Addr:Port  : 172.22.17.227:5060

Destn SIP Resp Addr:Port : 172.22.17.227:5060

Destination Name         : 172.22.17.227

Apr 20 16:53:35.099: //2159/33C3CE438E7D/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g711alaw

Negotiated Codec Bytes   : 160

Nego. Codec payload      : 8 (tx), 8 (rx)

Negotiated Dtmf-relay    : 6

Dtmf-relay Payload       : 100 (tx), 100 (rx)

Source IP Address (Media): 172.31.37.166

Source IP Port    (Media): 17772

Destn  IP Address (Media): 172.22.17.227

Destn  IP Port    (Media): 30014

Orig Destn IP Address:Port (Media): [ - ]:0

Apr 20 16:53:35.099: //2159/33C3CE438E7D/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 18

Disconnect Cause (SIP)   : 480

Regards and thanks

Everyone's tags (4)
2 ACCEPTED SOLUTIONS

Accepted Solutions
Hall of Fame Super Silver

calls CFA to voicemail fails from SIP Trunk

Can you please post "debug ccsip messages" from CUBE for one of these calls and your CUBE config?

What if you forward the call to another phone instead of VM, does that work?

Chris

Cisco Employee

Re: calls CFA to voicemail fails from SIP Trunk

++ Callmanager sent 503 service unavailable with cause value 47, "Resource unavailable". Looks like CCM is trying to enagage transcoder. Detailed CCM logs will reveal that. But can you check the  region relationship between CUBE sip trunk and VM sip trunk, and make sure to use g711alaw which  is negotiated by CUBE to CCM.

Apr 21 05:04:03.220: //2657/3F64E37891A4/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 10.101.105.2:5060;branch=z9hG4bK33A1E93

From: <973796965>;tag=2FE492D0-112D

To: <25785496>;tag=3588~ea7af75b-e2d3-45f8-8912-bca25030c3fd-37                                                                                        187566

Date: Sat, 21 Apr 2012 04:57:33 GMT

Call-ID: 3F661BC8-8AA611E1-91AAA1EB-323BBF7C@10.101.105.2

CSeq: 101 INVITE

Allow-Events: presence

Reason: Q.850;cause=47

Content-Length: 0

6 REPLIES
Hall of Fame Super Silver

calls CFA to voicemail fails from SIP Trunk

Can you please post "debug ccsip messages" from CUBE for one of these calls and your CUBE config?

What if you forward the call to another phone instead of VM, does that work?

Chris

New Member

Re: calls CFA to voicemail fails from SIP Trunk

Hi Chris thanks for the answer

i attached the debug ccsip messages for a call and cube config,

If forward the call to another phone the call from outside results without problems, the problem is only when users forwarded to VM.

Thanks and regards

Cisco Employee

Re: calls CFA to voicemail fails from SIP Trunk

++ Callmanager sent 503 service unavailable with cause value 47, "Resource unavailable". Looks like CCM is trying to enagage transcoder. Detailed CCM logs will reveal that. But can you check the  region relationship between CUBE sip trunk and VM sip trunk, and make sure to use g711alaw which  is negotiated by CUBE to CCM.

Apr 21 05:04:03.220: //2657/3F64E37891A4/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 10.101.105.2:5060;branch=z9hG4bK33A1E93

From: <973796965>;tag=2FE492D0-112D

To: <25785496>;tag=3588~ea7af75b-e2d3-45f8-8912-bca25030c3fd-37                                                                                        187566

Date: Sat, 21 Apr 2012 04:57:33 GMT

Call-ID: 3F661BC8-8AA611E1-91AAA1EB-323BBF7C@10.101.105.2

CSeq: 101 INVITE

Allow-Events: presence

Reason: Q.850;cause=47

Content-Length: 0

New Member

Re: calls CFA to voicemail fails from SIP Trunk

Thanks for the reply, both trunks are in the same region

Cisco Employee

Re: calls CFA to voicemail fails from SIP Trunk

Can you make a test call and provide detailed CCM traces  with the calling details ??

New Member

Re: calls CFA to voicemail fails from SIP Trunk

Thanks marmugam and Chris, your answers leads me to the solution, i checked the SDI log and i realize that i not putting the Media Resource Group on both trunks, i put this and the call flows like a charm.

Thanks again

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