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SIP Trunk ISP / One Way Audio Incoming calls

Hello all

We are having problems with one way audio from external calls to internal ramal.

The external phone hears the voice but the internal phone (extension) don't.

When internal phone makes a call to external number, the call works fine.

CUCM --> SIP Trunk CUBE --> ITSP

I attached the output from "debug ccsip message" and "debug voip rtp" but I separate this part in particular:

Mar 21 11:29:48.628: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:5099@189.86.188.178:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 189.23.0.181:5060;branch=z9hG4bKk15vn100008p1c2p1th0.1
Call-ID: 1n8ys842sbbc4cyyrs18by4cas2cabsc@SoftX3000
From: <sip:1136386615@189.23.0.181;user=phone>;tag=4de928y1-CC-29
To: <sip:5099@189.86.188.178;user=phone>
CSeq: 1 INVITE
Max-Forwards: 60
Contact: <sip:1136386615@189.23.0.181:5060;user=phone;transport=udp>
Min-SE: 90
Session-Expires: 1800
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R011
Supported: 100rel,timer
Content-Length: 350
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 46922623 46922623 IN IP4 189.23.1.229
s=Sip Call
c=IN IP4 189.23.1.229
t=0 0
m=audio 13350 RTP/AVP 18 8 118 110 100
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:118 PCMA/8000
a=gpmd:118 vbd=yes
a=rtpmap:110 PCMU/8000
a=gpmd:110 vbd=yes
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=fmtp:18 annexb=no

SIP: Trying to parse unsupported attribute at media level
SIP: Trying to parse unsupported attribute at media level

 

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Collaborator

Hi Leonardo,

Hi Leonardo,

From the logs attached, it seems like the CUBE receives an invite and sends trying, ringing and 200 OK but do not receive an ACK from the provider side. So its sending 200 OK multiple times and finally sends a BYE with cause code 86 - Network timeout.

HTH

Rajan

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