SIP to SIP external calls voice mail not working

[reda.man]

Hi All

I have a CME on 3925 running ISO 15.5, and the CUE is 8.6 from sip trunk to CUE Voicemail.

the call comes in after 2 rings it should go to voicemail, but the call drops for a sec then it carries on ringing again.

no supplementary-service sip moved-temporarily
no supplementary-service sip refer

already configured

Thank you in advance.

debug CCSIP messages:

Oct 16 17:13:47.758: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0208859@51.52.XX.XX;user=phone SIP/2.0
Max-Forwards: 67
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: 100rel,timer
To: <sip:0208859@88.215.XX.XX;user=phone>
From: <sip:079XXXXXXXX@88.215.XX.XX;user=phone>;tag=3780238407-1185467282
P-Asserted-Identity: <sip:079XXXXXXXX@88.215.XX.XX;user=phone>
Call-ID: 124779881-3780238407-528216619@
CSeq: 1 INVITE
Allow: UPDATE,PRACK,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Via: SIP/2.0/UDP 88.215.XX.XX:5060;branch=z9hG4bKc7540e5053ac4ed3782c1fc98117e7bf
Contact: <sip:079XXXXXXXX@88.215.XX.XX:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 265

v=0
o=MSX24 319246336 1571249608 IN IP4 88.215.XX.XX
s=sip call
c=IN IP4 88.215.XX.XX
t=0 0
m=audio 30458 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

Oct 16 17:13:47.762: //12494/267C094DB3AC/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 88.215.XX.XX:5060;branch=z9hG4bKc7540e5053ac4ed3782c1fc98117e7bf
From: <sip:079XXXXXXXX@88.215.XX.XX;user=phone>;tag=3780238407-1185467282
To: <sip:0208XXXXXXX@88.215.XX.XX;user=phone>

IP/2.0 180 Ringing
Via: SIP/2.0/UDP 88.215.XX.XX:5060;branch=z9hG4bKc7540e5053ac4ed3782c1fc98117e7bf
From: <sip:079XXXXXXXX@88.215.XX.XX;user=phone>;tag=3780238407-1185467282
To: <sip:0208XXXXXXX@88.215.XX.XX;user=phone>;tag=297B667C-1A4D
Date: Wed, 16 Oct 2019 19:13:47 GMT

CSeq: 1 INVITE
Require: 100rel
RSeq: 6111
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event

[Rajan]

Hi Reda,

Can you share the below debugs and config for one such call to check

debug voip ccapi inout
debug ccsip messages

[reda.man]

Hi Rajan

Thank you for your prompt reply, I was away yesterday.

I will post the debug output for ccsip message, as well as debug voice ccapi inout 

Thanks in advance

Reda

[reda.man]

Hi Rajan

any update on this please?

Thank you.

Reda

[Rajan]

Hi Reda,

Sorry, I was busy with other work. I looked at the debugs and could see the call failing with cause code - 47 media resource unavailable.

Warning: 399 51.X.X.X "Transcoder Not Configured"
Server: Cisco-SIPGateway/IOS-15.5.3.M6a
Reason: Q.850;cause=47

I could see the incoming call using G711alaw codec. Could you please check whether you are using G711ulaw for voicemail. You need to make sure that same codec is used throughout the call flow (till voicemail system) or configure transcoders to use for these calls.

HTH
Rajan
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[reda.man]

Thank you Rajan for the reply

the codec for VM is g711ulaw, and I created a voice-class codec that has most of codecs, and now I am getting an error messenger destination unreachable.

I will run both debugs again I will post them later

Thanks again for your help