06-09-2015 11:58 AM - edited 03-17-2019 03:16 AM
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
06-09-2015 12:46 PM
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
06-09-2015 01:39 PM
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
06-09-2015 01:56 PM
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
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