[cisco-voip] [Bulk]redirect Voicemail to CUE from external SIP number

jaystants@rogers.com jaystants at rogers.com
Thu Jul 28 13:22:49 EDT 2011


What specific debugs should be run ? 

Sent from my HTC bolt of lightning...

----- Reply message -----
From: "ccieid1ot" <ccieid1ot at gmail.com>
To: "Jay Stants" <jaystants at rogers.com>
Cc: "Peter Slow" <peter.slow at gmail.com>, "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: [cisco-voip] [Bulk]redirect Voicemail to CUE from external SIP number
Date: Thu, Jul 28, 2011 10:53 am
Run some debugs to see what's the redirecting number when it's forwarding to voice mail.  Might need a redirecting translation

On Tue, Jul 26, 2011 at 11:30 AM, Jay Stants <jaystants at rogers.com> wrote:


Peter,
 
I have the following within my config
 
no voice register global
 
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

 
Regards,Jay Stants
jaystants at rogers.com







From: Peter Slow <peter.slow at gmail.com>
To: "jaystants at rogers.com" <jaystants at rogers.com>

Cc: cisco-voip at puck.nether.net
Sent: Monday, July 25, 2011 5:30:45 AM

Subject: Re: [cisco-voip] [Bulk]redirect Voicemail to CUE from external SIP number

your sample config does not have

voice service voi

  allow s t s

under it. you need the sip to sip command to
allow it to function as a
CUBE in your case, since we're connecting two SIP dial-peers together.

Is that done?

-Peter

On Fri, Jul 22, 2011 at 4:30 PM, jaystants at rogers.com

<jaystants at rogers.com> wrote:
> Any one able to assist with this issue?
>
> Regards,
>
> Jay Stants
>
> Sent from my Thunderbolt 4G LTE

>
>
> ----- Reply message -----
> From: "Jay Stants" <jaystants at rogers.com>
> To: <cisco-voip at puck.nether.net>

> Subject: [Bulk] [cisco-voip] redirect Voicemail to CUE from
external SIP
> number
> Date: Thu, Jul 21, 2011 8:41 am
>
>
> I've managed to get VM working internally on test phones but when i dial in
> from my DID, call gets redirected to a specific phone (which is the

> behaviour i want) but after timeout duration call is not transfered to
> voicemail, instead just rings busy . Can someone point out anything i may be
> missing (translation or dial-peer or possibly something else)

>
> Help is much appreciated
>
> Config Details
> ---------------
> snet-wan2#sh run
> Building configuration.....
> Current configuration : 6793 bytes
> !
> ! Last configuration change at 08:17:38 EST Thu Jul 21 2011 by netmgmt

> ! NVRAM config last updated at 08:17:38 EST Thu Jul 21 2011 by netmgmt
> !
> version 15.1
> 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
> !
> !
> sip-ua
>  credentials username {removed} password 7 {removed} realm newyork.voip.ms

>  authentication username {removed} password 7 {removed}
>  no remote-party-id
>  retry invite 2
>  retry register 10
>  timers connect 100
>  mwi-server ipv4:1.1.1.2 expires 3600 port 5060 transport tcp unsolicited

>  registrar dns:newyork.voip.ms expires 180
>  sip-server dns:newyork.voip.ms
>
 host-registrar
> !
> !
> !
> telephony-service
>  authentication credential {username password}
>  max-ephones 15
>  max-dn 15
>  ip source-address 10.50.1.2 port 2000

>  system message Cisco CME 8.1
>  url services http://1.1.1.2/voiceview/common/login.do
>  url authentication http://1.1.1.1/CCMCIP/authenticate.asp

>  cnf-file location flash:
>  cnf-file perphone
>  time-zone 12
>  dialplan-pattern 1 5856786019 extension-length 4
>  voicemail 4000

>  max-conferences 8 gain -6
>  dn-webedit
>  time-webedit
>  transfer-system full-consult
>  secondary-dialtone 9
>  create cnf-files
version-stamp Jan 01 2002 00:00:00
> !
> !
> 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
> !
> !
> ephone-dn  13
> !
> !
> ephone-dn  14
>  number 8000...........
>  mwi on
> !
> !

> ephone-dn  15
>  number 8001...........
>  mwi off
> !
> !
> ephone  1
>  device-security-mode none
>  mac-address 001E.7AC5.896A
>  username "user" password {removed}

>  type 7961GE
>  button  1:1
>
>
> Regards,
> Jay Stants
> jaystants at rogers.com
>
> _______________________________________________

> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

>
>



_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net

https://puck.nether.net/mailman/listinfo/cisco-voip




-- 
duy
CCIE #27737 Voice
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20110728/b4839a8c/attachment.html>


More information about the cisco-voip mailing list