[cisco-voip] Calls not forwarding over SIP trunk to Cue

Sat Aug 13 23:24:04 EDT 2011

debug ccsip messages output
 
xx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:5856786019 at 74.74.255.254:56665 SIP/2.0
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>
Contact: <sip:5856784306 at 74.63.41.218>
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
CSeq: 102 INVITE
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "+15856784306" <sip:5856784306 at 74.63.41.218>;privacy=off;screen=no
Date: Sun, 14 Aug 2011 03:21:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 2851 2851 IN IP4 74.63.41.218
s=session
c=IN IP4 74.63.41.218
t=0 0
m=audio 14492 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Aug 14 03:21:35.836: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>
Date: Sun, 14 Aug 2011 03:21:35 GMT
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Aug 14 03:21:35.836: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>;tag=64143B74-13D4
Date: Sun, 14 Aug 2011 03:21:35 GMT
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:5856786019 at 74.74.255.254:5060>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

Aug 14 03:21:47.324: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:5856786019 at 74.74.255.254:56665 SIP/2.0
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
CSeq: 102 CANCEL
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "+15856784306" <sip:5856784306 at 74.63.41.218>;privacy=off;screen=no
Content-Length: 0

Aug 14 03:21:47.356: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>
Date: Sun, 14 Aug 2011 03:21:47 GMT
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
CSeq: 102 CANCEL
Content-Length: 0

Aug 14 03:21:47.356: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>;tag=64143B74-13D4
Date: Sun, 14 Aug 2011 03:21:47 GMT
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0

Aug 14 03:21:47.400: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:5856786019 at 74.74.255.254:56665 SIP/2.0
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>;tag=64143B74-13D4
Contact: <sip:5856784306 at 74.63.41.218>
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
CSeq: 102 ACK
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "+15856784306" <sip:5856784306 at 74.63.41.218>;privacy=off;screen=no
Content-Length: 0

 
Regards,
Jay Stants
jaystants at rogers.com


From: "Buchanan, James" <jbuchanan at presidio.com>
>To: Jay Stants <jaystants at rogers.com>; "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
>Sent: Saturday, August 13, 2011 11:11:50 PM
>Subject: RE: [cisco-voip] Calls not forwarding over SIP trunk to Cue
>
>
>Can you send a debug ccsip messages?
> 
>James Buchanan| UC Technology Manager |Presidio South |Presidio Networked Solutions 
>12 Cadillac Dr Ste 130 Brentwood, TN 37027 |jbuchanan at presidio.com
>D: 615-866-5729 |F:615-866-5781  www.presidio.com
> 
>From:Jay Stants [mailto:jaystants at rogers.com] 
>Sent: Saturday, August 13, 2011 10:02 PM
>To: Buchanan, James; cisco-voip at puck.nether.net
>Subject: Re: [cisco-voip] Calls not forwarding over SIP trunk to Cue
> 
>no dice - still just rings infinantly .. 
>
>
> 
>Regards,
>Jay Stants
>jaystants at rogers.com
>From:"Buchanan, James" <jbuchanan at presidio.com>
>To: Jay Stants <jaystants at rogers.com>; "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
>Sent: Saturday, August 13, 2011 10:55:22 PM
>Subject: RE: [cisco-voip] Calls not forwarding over SIP trunk to Cue
>Try adding b2bua onto your voicemail dial peer. 
> 
>James Buchanan| UC Technology Manager |Presidio South |Presidio Networked Solutions 
>12 Cadillac Dr Ste 130 Brentwood, TN 37027 |jbuchanan at presidio.com
>D: 615-866-5729 |F:615-866-5781  www.presidio.com
> 
>From:cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jay Stants
>Sent: Saturday, August 13, 2011 9:49 PM
>To: cisco-voip at puck.nether.net
>Subject: [cisco-voip] Calls not forwarding over SIP trunk to Cue
> 
>Having an issue where calls coming in from ITSP ring and never fwd to voicemail. Internally voicemail works. either by dialing the vm number or by pushing the messages button which essentially just speed dials the 4000 extension used for voicemail. Can someone take a look at the included config and maybe point out what i'm missing to allow for calls to be fwd'ed to Cue after the specified time.
> 
>voice service voip
> ip address trusted list
>  ipv4 0.0.0.0 0.0.0.0
> allow-connections h323 to h323
> allow-connections h323 to sip
> allow-connections sip to h323
> allow-connections sip to sip
> supplementary-service h450.12
> sip
>  registrar server expires max 600 min 60
>!
>!
>!
>!
>voice translation-rule 1
> rule 1 /^9/ //
>!
>voice translation-rule 3
> rule 1 /4.../ /5856786019/
>!
>!
>voice translation-profile voip.ms
> translate calling 3
> translate called 1
>dial-peer voice 1 voip
> description **SIP Trunk to newyork.voip.ms**
> translation-profile outgoing voip.ms
> destination-pattern 9[2-9].[2-9].......
> session protocol sipv2
> session target dns:newyork.voip.ms
> dtmf-relay rtp-nte sip-notify
> codec g711ulaw
> no vad
>!
>dial-peer voice 2 voip
> description **Incoming SIP Trunk - Voip.ms**
> translation-profile incoming voip.ms
> session protocol sipv2
> session target ipv4:10.50.1.2
> incoming called-number 5856786019
> dtmf-relay sip-notify
> codec g711ulaw
> no vad
>!
>dial-peer voice 20 pots
> destination-pattern 5.T
> direct-inward-dial
> no sip-register
>!
>dial-peer voice 3 voip
> description ** Voicemail **
> destination-pattern 4000
> session protocol sipv2
> session target ipv4:1.1.1.2
> dtmf-relay sip-notify
> codec g711ulaw
> no vad
>    
>ephone-dn-template  1
> call-forward busy 4000
> call-forward noan 4000 timeout 18
> 
>ephone-dn  1  dual-line
> number 4005 secondary 5856786019 no-reg
> label 4005
> name Jay Stants
> ephone-dn-template 1
> 
> 
>Regards,
>Jay Stants
>jaystants at rogers.com
> 
>
>
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