10-17-2019 11:12 AM
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
10-18-2019 07:25 AM
10-19-2019 03:50 AM
11-09-2019 04:48 AM
Hi Rajan
any update on this please?
Thank you.
Reda
11-14-2019 10:20 PM
11-15-2019 03:43 AM
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
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