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Cue issue with forward to voicemail.

Hello,

I have a issue with CME + CUE when forwarding calls to voicemail.

When receive an incoming call from SIP Provider number, the correct phone associated starts ringing. After 10 seconds the phone forwards the call to the CUE voicemail pilot number, but who calling hear ringing endless. Seems that CUE is not able to respond.

I'm sure that the matter is the process of forwarding associated to incoming SIP calls, because I have tried this:

incoming call from PSTN is forwarded correctly to voicemail.

incoming call from SIP directed to CUE voicemail (VM administration number from PSTN) is successful (I can hear Cue asking my ID).

Thank you for help

8 Replies 8

MMstre
Level 3
Level 3

Please attach CME and CUE config.

Here is the output of "debug ccsip messages" of an inbound call from SIP Provider "voip.eutelia.it".

Oct 14 14:34:44.950: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bKb45b.e97348b5.0,SIP/2.0/UDP

62.94.71.98:5060;rport=61765;x-route-tag="tgrp:Slot6";branch=z9hG4bK5AD745ECB

From: <>;tag=D776D9A4-71A

To: <>;tag=29657134-1907

Date: Wed, 14 Oct 2009 14:34:44 GMT

Call-ID: 8D09ACD7-B80511DE-87D591AB-C6312517@62.94.71.98

CSeq: 101 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO,

REGISTER

Allow-Events: telephone-event

Remote-Party-ID: "Gianluigi" <340>;party=called;screen=no;privacy=off

Contact: <>

Record-Route: <83.211.227.21>

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Oct 14 14:34:54.970: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 302 Moved Temporarily

Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bKb45b.e97348b5.0,SIP/2.0/UDP

62.94.71.98:5060;rport=61765;x-route-tag="tgrp:Slot6";branch=z9hG4bK5AD745ECB

From: <>;tag=D776D9A4-71A

To: <>;tag=29657134-1907

Date: Wed, 14 Oct 2009 14:34:44 GMT

Call-ID: 8D09ACD7-B80511DE-87D591AB-C6312517@62.94.71.98

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Diversion: <>;reason=no-answer;counter=1

Contact: <>

Content-Length: 0

Oct 14 14:34:55.018: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:***IDCALLED***@192.168.0.249:5060 SIP/2.0

Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bKb45b.e97348b5.0

From: <>;tag=D776D9A4-71A

Call-ID: 8D09ACD7-B80511DE-87D591AB-C6312517@62.94.71.98

To: <>;tag=29657134-1907

CSeq: 101 ACK

User-Agent: SPS EUT RM GW 02 (0.9.6 (i386/linux))

Content-Length: 0

At this point the call keeps ringing for the caller and not forwarded to voicemail.

In attach you will find the cme and cue configuration.

Thanks

Try

voice service voip

no supplementary-service sip moved-temporarily

I've tried with this result:

Oct 14 16:28:49.511: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 501 Not Implemented

Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bK852d.09e461c5.0,SIP/2.0/UDP 83.211.2.218:5060;rport=58994;x-route-tag="tgrp:Slot7";branch=z9hG4bK4FB8A312A5

From: <>;tag=3DB5A9A8-A20

To: <>;tag=29CDBA24-1956

Date: Wed, 14 Oct 2009 16:28:39 GMT

Call-ID: 76C2D8AF-B81511DE-8CAD92F2-3F8F6970@83.211.2.218

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=65

Content-Length: 0

Oct 14 16:28:49.555: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:***IDCALLED***@192.168.0.249:5060 SIP/2.0

Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bK852d.09e461c5.0

From: <>;tag=3DB5A9A8-A20

Call-ID: 76C2D8AF-B81511DE-8CAD92F2-3F8F6970@83.211.2.218

To: <>;tag=29CDBA24-1956

CSeq: 101 ACK

User-Agent: SPS EUT RM GW 03 (0.9.6 (i386/linux))

Content-Length: 0

The caller hear Error Tone and is not forwarded to Voicemail.

I am assuming that your voicemail pilot is 6100. Try entering these commands in the voicemail dial-peer:

mailbox-selection orig-called-num

voice-class sip outbound-proxy ipv4:

The problem may be that that the number it is forwarding to is being sent back to the registrar, who does not recognize the number, assuming that 6100 is not a DID. Who knows what they do with it. By entering the command voice-class sip outbound-proxy in the dial-peer, it forces the number to be redirected to the CUE module.

Sorry, forgot to ask. Is your service provider acting as an outbound proxy as well?

I've applied your suggested command, but with no result.

here the debug ccsip:

Oct 15 08:38:41.267: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bK2f31.66c0b2a6.0,SIP/2.0/UDP 83.211.2.216:5060;rport=56053;x-route-tag="tgrp:Slot6";branch=z9hG4bK6517E9252B

From: <>;tag=C692248C-EF

To: <>;tag=2D45CD58-1E63

Date: Thu, 15 Oct 2009 08:38:41 GMT

Call-ID: F9B3A0E7-B89C11DE-8D3EBF13-D44B072@83.211.2.216

CSeq: 101 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: "Gianluigi" <340>;party=called;screen=no;privacy=off

Contact: <>

Record-Route: <83.211.227.21>

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Oct 15 08:38:51.279: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 302 Moved Temporarily

Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bK2f31.66c0b2a6.0,SIP/2.0/UDP 83.211.2.216:5060;rport=56053;x-route-tag="tgrp:Slot6";branch=z9hG4bK6517E9252B

From: <>;tag=C692248C-EF

To: <>;tag=2D45CD58-1E63

Date: Thu, 15 Oct 2009 08:38:41 GMT

Call-ID: F9B3A0E7-B89C11DE-8D3EBF13-D44B072@83.211.2.216

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Diversion: <>;reason=no-answer;counter=1

Contact: <6100>

Content-Length: 0

Oct 15 08:38:51.327: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:***IDCALLED***@192.168.0.249:5060 SIP/2.0

Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bK2f31.66c0b2a6.0

From: <>;tag=C692248C-EF

Call-ID: F9B3A0E7-B89C11DE-8D3EBF13-D44B072@83.211.2.216

To: <>;tag=2D45CD58-1E63

CSeq: 101 ACK

User-Agent: SPS EUT RM GW 01 (0.9.6 (i386/linux))

Content-Length: 0

I don't know if my provider act as an outbound proxy.

What I think is:

When I recieve a call from PSTN the CME forwards correctly the call to Voicemail. Instead, during an incoming SIP call the CME leaves the forwarding process to Service Provider.

There is any way to force the SIP forwarding process to working only locally?

Thanks for this post I've been struggling with this issue of the global outbound-proxy command for ages.

 

I've applied "voice-class sip outbound-proxy ipv4:" to my voicemail DP and it has kept the call local even when "session target ipv4:" existed and did not help.

Though "mailbox-selection orig-called-num" did not help and I'll have to find a way to preserve the called num/ext as it doesn't go to the appropriate mailbox when <messages> is hit or when outside calls go to voicemail the call is terminated as there is no specified mailbox.

 

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