10-13-2009 02:07 AM - edited 03-15-2019 08:02 PM
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
10-14-2009 05:26 AM
Please attach CME and CUE config.
10-14-2009 08:18 AM
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=off340>
Contact: <>>
Record-Route: <83.211.227.21>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
10-14-2009 08:23 AM
Try
voice service voip
no supplementary-service sip moved-temporarily
10-14-2009 08:38 AM
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.
10-14-2009 09:18 AM
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.
10-14-2009 09:19 AM
Sorry, forgot to ask. Is your service provider acting as an outbound proxy as well?
10-15-2009 01:01 AM
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=off340>
Contact: <>>
Record-Route: <83.211.227.21>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>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?
08-18-2015 12:53 AM
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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