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 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>;tag=ryFQea59e6Upe

   To: <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>

   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>;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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