11-08-2016 07:12 PM - edited 03-17-2019 08:37 AM
Hi Guys
I´ve a CME with SIP Trunk to ITSP and registered as well. My issue is that, incoming calls are not working, per debug I´m receiving:
Nov 1 11:59:34.670: //46650/7D711B609860/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 162.243.227.119;branch=z9hG4bKe54.5c64ba90b0454be6c61f76d2eaf7ad19.0,SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK72222d00;rport=5080
From: "11992454948" <sip:11992454948@127.0.0.1:5080>;tag=as7644e2cc
To: <sip:1000_pabx1691.ligou.me@200. 10.13.9:5060>;tag=18468E48-135B
Date: Tue, 01 Nov 2016 11:59:34 GMT
Call-ID: 66feb62f1b8bcaa64a87939a5e0ac6cc@127.0.0.1:5080
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=28
Content-Length: 0
Any help will be appreciated, see the attached files.
Solved! Go to Solution.
11-11-2016 12:49 AM
I do not see here translation rules.
so incoming calling number must be converted to numbers 1000 or 1001
and
voice class sip-profiles 10
is not applied to incoming dial-peer
11-11-2016 12:34 AM
are you really need IPv6 ?
ipv6 unicast-routing ipv6 cef
11-11-2016 05:55 AM
No, I do not !!
11-11-2016 12:49 AM
I do not see here translation rules.
so incoming calling number must be converted to numbers 1000 or 1001
and
voice class sip-profiles 10
is not applied to incoming dial-peer
11-11-2016 05:54 AM
Thanks for your reply (+5)
I can`t XLATE the URI 1000_pabx1691.ligou.me using translation rule, right? Just created the Voice Class SIP-PROFILE to do it, applied to voice service voip but did not worked.
I`ll try apply in the incoming DP and let you know!!!
11-18-2016 06:54 AM
I was unable to do that test, spoke with ITSP guys and they changed the information sent to our CME:
Changed form:
To: <sip:1000_pabx1691.ligou.me@200. 10.13.9:5060>;tag=18468E48-135B
Changed to:
To: <sip:1000@200. 10.13.9:5060>;tag=18468E48-135B
Thanks for your help, I´ll mark this discussion as answered
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