m=audio 18646 RTP/AVP 0 8 101 13 ^ ^ - these are codecs
Hi,
My outbound SIP invites are going out without a codec list? Could someone point me in the right direction on how to set / insert this?
send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565:
------------------------------------------------------------------------ INVITE sip:0800000000@1.2.3.4:5060 <http://sip:0800000000@1.2.3.4:5060> SIP/2.0 Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr Max-Forwards: 69 From: "Liam - Test" <sip:12345678@9.8.7.6 <mailto:sip%3A12345678@9.8.7.6>>;tag=ryFQea59e6Upe To: <sip:0800000000@1.2.3.4:5060 <http://sip:0800000000@1.2.3.4:5060>> Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43 CSeq: 12805602 INVITE Contact: <sip:12345678@9.8.7.6:5060 <http://sip:12345678@9.8.7.6:5060>> User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 189 X-FS-Support: update_display Remote-Party-ID: "Liam - Test" <sip:12345678@9.8.7.6 <mailto:sip%3A12345678@9.8.7.6>>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6 s=FreeSWITCH c=IN IP4 9.8.7.6 t=0 0 m=audio 18646 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16
------------------------------------------------------------------------
Thanks
Liam
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