[Freeswitch-users] Outbound SIP invites without codec list

["Корнев Е.С."]

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 at 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 at 9.8.7.6 
> <mailto:sip%3A12345678 at 9.8.7.6>>;tag=ryFQea59e6Upe
>    To: <sip:0800000000 at 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 at 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 at 9.8.7.6 
> <mailto:sip%3A12345678 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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-- 



С уважением, Корнев Е.С.
Руководитель группы СПГИ
ОАО "СЦК"
(343) 378-3100 + 170



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