02-03-2020 11:13 PM - edited 02-03-2020 11:17 PM
Facing issue with Vocus SIP trunk incoming call. Receiving invite but not routing or matching dial peer. Attached the config please do the needful.
Received:
INVITE sip:0380809216@10.224.78.34:5060 SIP/2.0
Via: SIP/2.0/UDP 202.147.134.21:5060;branch=z9hG4bK3nglve00e84ita1er0e0.1
From: <sip:0919739994990@amcomvoice.ipsystems.com.au;user=phone>;tag=SD0defd01-266667079-1580797966377-
To: "380809216 380809216"<sip:0380809216@amcomvoice.ipsystems.com.au>
Call-ID: SD0defd01-d5c3a28df4d9048c86c0153af2f2d89a-jm6gpa0030
CSeq: 125000725 INVITE
Contact: <sip:SDl5i79-vp9pm6rhjjonlr5pmpv6hvrkbk0qgvagofge3pvv404m90c42sc0@202.147.134.21:5060;transport=udp>
P-Called-Party-ID: <sip:0380809216@amcomvoice.ipsystems.com.au>
Supported: 100rel,timer
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Recv-Info: x-broadworks-client-session-info
Accept: application/media_control+xml,application/sdp,multipart/mixed
Min-SE: 90
Session-Expires: 1800;refresher=uas
Max-Forwards: 29
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 270
v=0
o=BroadWorks 5479777217 1 IN IP4 202.147.134.21
s=-
c=IN IP4 202.147.134.21
t=0 0
m=audio 18752 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=bsoft: 1 image udptl t38
02-04-2020 05:36 AM
Have you run a debug to check dial peer matching "debug voip dialpeer inout"? I'm wondering if it's not matching the correct inbound dial peer, and therefore not getting translated.
05-25-2021 09:07 AM
Hi
How did you fix your problems?
I have the same issue.
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