cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
456
Views
0
Helpful
3
Replies

CME SIP inbound call Redirect to VM in cloud problem

ciscotac16
Level 1
Level 1

Hello!

I rarely work on CME, and so would greatly appreciate the assistance of a local wizard to work out how to implement a "redirect" of an inbound call from my SIP provider to another number (VM in the cloud) if the called number isn't answered.  I don't really want to do this on ephones, btw, since the number in question is not the primary number on the ephones, and I want it to go to a single VM box in the cloud, irrespective of individual phones.  I'm running CME 8.6.  Applicable parts of Inbound and Outbound configs are below. 

Thanks much!

Deb

=======================================================

NOTE:  All DNs and SIP client codes changed to bogus numbers.  So, using these numbers, I want calls that come in for 7075551111 to be redirected back out to 7075552222. 

 

INBOUND:


voice translation-profile SIPin-Prefix9
 translate calling 13
voice translation-rule 13
 rule 1 /\(^.......$\)/ /9\1/
 rule 2 /\(^..........$\)/ /91\1/
 rule 3 /\(^1..........$\)/ /9\1/

dial-peer voice 7 voip
 description "Incoming Call from SIP Trunk"
 translation-profile incoming SIPin-Prefix9
 preference 2
 session protocol sipv2
 session target dns:sip.provider.com
 incoming called-number 17075551111
 voice-class codec 1  
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
!

OUTBOUND:

voice translation-profile SIPout-LD
 translate calling 15
 translate called 8
 translate redirect-target 6
 translate redirect-called 6
voice translation-rule 15
 rule 1 /.*/ /17075551111/
voice translation-rule 8
 rule 1 /^9\(1[2-9]..[2-9]......\)/ /333333333*\1/
voice translation-rule 6
 rule 1 /^9\(1[2-9]..[2-9]......\)/ /333333333*17075552222\1/

dial-peer voice 2 voip
 description **Outgoing call to LD numbers via SIP**
 translation-profile outgoing SIPout-LD
 preference 2
 destination-pattern ^91[2-9]..[2-9]......$
 session protocol sipv2
 session target dns:sip.provider.com
 voice-class codec 1  
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 clid restrict
 no vad

3 Replies 3

yahsiel2004
Level 7
Level 7

Can you perform a "debug ccsip message"? Also can you post your "voip service voip" config. I have a feeling you will need to perform the following.

voice service voip

no supplementary-service sip moved-temporarly

HTH

Yosh
 

HTH Regards, Yosh

Hi, Yosh.

Here is the "voip service voip" config:

voice service voip
 ip address trusted list
  ipv4 <address removed>
  ipv4 <address removed>
 allow-connections sip to sip
 sip
  bind control source-interface FastEthernet0/0
  bind media source-interface FastEthernet0/0
  session transport tcp
  registrar server expires max 3600 min 3600
  transport switch udp tcp
  sip-profiles 100
  no call service stop
!

I'm attaching a file with the debug output.

 

Thank you for your time and suggestions!

Deb

I think your issue is that you're sending a number that doesn't exist "333333333*17075551111". I see what you're trying to do but I don't think that this is the way to go about it. I'll research this further and if I find something I will let you know. Maybe someone here can help you out.

Regards,

Yosh

HTH Regards, Yosh