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

jaystants@rogers.com jaystants at rogers.com
Sat Aug 13 23:16:03 EDT 2011


Will send in a few..have to capture some calls first.

Sent from my HTC bolt of lightning...

----- Reply message -----
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>
Subject: [cisco-voip] Calls not forwarding over SIP trunk to Cue
Date: Sat, Aug 13, 2011 11:11 pm
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.comD: 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.comFrom: "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 CueTry 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.comD: 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 1dial-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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