[Freeswitch-users] SPA921 Problem Bad Session Description

[Brian West]

The SDP is invalid... I suspect your PAP2 has a properly formatted SDP... Can you not see on the two lines for "a=rtpmap:8 /8000" and "a=rtpmap:0 /8000" are the invalid lines they should be "a=rtpmap:8 PCMA/8000" and "a=rtpmap:0 PCMU/8000" .  We WILL NOT make this work because its invalid.  Its on you to contact the manufacture of your device and have them fix it.

/b

On Oct 18, 2010, at 5:32 AM, Maxim Balabaev wrote:

> I can`t make calls from linksys spa921 because of "400 Bad Session Description", incoming are ok. pap2t works perfectly. freeswitch is rev from git trunk. Here comes logs:
>    ------------------------------------------------------------------------
> recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
>    ------------------------------------------------------------------------
>    INVITE sip:1001 at xxx SIP/2.0
>    Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport
>    From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
>    To: <sip:1001 at xxx>
>    Call-ID: 14d793ed-1abf9d0e at xxx
>    CSeq: 101 INVITE
>    Max-Forwards: 70
>    Contact: <sip:022 at xxx:5060>
>    Expires: 240
>    User-Agent: Linksys/SPA921-5.1.8
>    Content-Length: 386
>    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>    Supported: replaces
>    Content-Type: application/sdp
>    
>    v=0
>    o=- 603544 603544 IN IP4 xxx
>    s=-
>    c=IN IP4 xxx
>    t=0 0
>    m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
>    a=rtpmap:8 /8000
>    a=rtpmap:0 /8000
>    a=rtpmap:2 G726-32/8000
>    a=rtpmap:4 G723/8000
>    a=rtpmap:18 G729/8000
>    a=rtpmap:96 G726-40/8000
>    a=rtpmap:97 G726-24/8000
>    a=rtpmap:98 G726-16/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-15
>    a=ptime:30
>    a=sendrecv
>    ------------------------------------------------------------------------
> send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
>    ------------------------------------------------------------------------
>    SIP/2.0 400 Bad Session Description
>    Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060
>    From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
>    To: <sip:1001 at xxx>;tag=ytUD271ypvy6r
>    Call-ID: 14d793ed-1abf9d0e at xxx
>    CSeq: 101 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16 -0400
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Content-Length: 0
>   
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