Hello, i know it sounds weird, but just for 1 customer, a2billing is not sending g729 codec as an option, the thing is that the client wants to use g729 only, i recheck my installation and test with other clients and g729 codec works just fine, but not with this one, i made a sip set debug on, and i found this:
Audio is at 17426
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 1xx.2xx.1xx.2xx:5060:
INVITE sip:
[email protected] SIP/2.0
Via: SIP/2.0/UDP 19x.2xx.2.2xx:5060;branch=z9hG4bK2f2e8fe0;rport
Max-Forwards: 70
From: "52" <sip:
[email protected]>;tag=as6d3de7d5
To: <sip:
[email protected]>
Contact: <sip:
[email protected]:5060>
Call-ID:
[email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.7.0)
Date: Wed, 18 Feb 2015 17:18:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Diversion: <sip:
[email protected]>;reason=unconditional
Content-Type: application/sdp
Content-Length: 288
v=0
o=root 2042696856 2042696856 IN IP4 1xx.2xx.1xx.2xx
s=Asterisk PBX 11.7.0
c=IN IP4 1xx.2xxx.1xxx.2xx
t=0 0
m=audio 17426 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv