Problem was that fields with some codecs names were blank. Thats why there were <meta charset="utf-8">"a=rtpmap:0 /8000". Also FS don`t have aliase betwen PCMA and G711.a for ex.<div>>> I'm not sure, but 5.1.8 firmware on linksys is quite outdated, isn't it?<br>
It`s really old but this is last one on their website.</div><div><br><div class="gmail_quote">2010/10/19 Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
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.<br>
<br>
/b<br>
<div><div></div><div class="h5"><br>
On Oct 18, 2010, at 5:32 AM, Maxim Balabaev wrote:<br>
<br>
> 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:<br>
> ------------------------------------------------------------------------<br>
> recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:<br>
> ------------------------------------------------------------------------<br>
> INVITE sip:1001@xxx SIP/2.0<br>
> Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport<br>
> From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0<br>
> To: <sip:1001@xxx><br>
> Call-ID: 14d793ed-1abf9d0e@xxx<br>
> CSeq: 101 INVITE<br>
> Max-Forwards: 70<br>
> Contact: <sip:022@xxx:5060><br>
> Expires: 240<br>
> User-Agent: Linksys/SPA921-5.1.8<br>
> Content-Length: 386<br>
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER<br>
> Supported: replaces<br>
> Content-Type: application/sdp<br>
><br>
> v=0<br>
> o=- 603544 603544 IN IP4 xxx<br>
> s=-<br>
> c=IN IP4 xxx<br>
> t=0 0<br>
> m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101<br>
> a=rtpmap:8 /8000<br>
> a=rtpmap:0 /8000<br>
> a=rtpmap:2 G726-32/8000<br>
> a=rtpmap:4 G723/8000<br>
> a=rtpmap:18 G729/8000<br>
> a=rtpmap:96 G726-40/8000<br>
> a=rtpmap:97 G726-24/8000<br>
> a=rtpmap:98 G726-16/8000<br>
> a=rtpmap:101 telephone-event/8000<br>
> a=fmtp:101 0-15<br>
> a=ptime:30<br>
> a=sendrecv<br>
> ------------------------------------------------------------------------<br>
> send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:<br>
> ------------------------------------------------------------------------<br>
> SIP/2.0 400 Bad Session Description<br>
> Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060<br>
> From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0<br>
> To: <sip:1001@xxx>;tag=ytUD271ypvy6r<br>
> Call-ID: 14d793ed-1abf9d0e@xxx<br>
> CSeq: 101 INVITE<br>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16 -0400<br>
> Accept: application/sdp<br>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY<br>
> Supported: timer, precondition, path, replaces<br>
> Allow-Events: talk, hold, refer<br>
> Content-Length: 0<br>
><br>
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