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

Buchanan, James jbuchanan at presidio.com
Sat Aug 13 23:11:50 EDT 2011


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<mailto:jbuchanan at presidio.com>
D: 615-866-5729 | F:615-866-5781  www.presidio.com<http://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<mailto: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<mailto:jbuchanan at presidio.com>
D: 615-866-5729 | F:615-866-5781  www.presidio.com<http://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<mailto:jaystants at rogers.com>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20110813/9ef07513/attachment.html>


More information about the cisco-voip mailing list