[Freeswitch-users] Outbound SIP invites without codec list

[Liam Farr]

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 at 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 at 9.8.7.6>;tag=ryFQea59e6Upe
   To: <sip:0800000000 at 1.2.3.4:5060>
   Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43
   CSeq: 12805602 INVITE
   Contact: <sip:12345678 at 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 at 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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