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<DIV dir=ltr align=left><SPAN class=977005513-19102010><FONT color=#0000ff
size=2 face=Arial>I'm not sure, but 5.1.8 firmware on linksys is
quite outdated, isn't it?</FONT></SPAN></DIV>
<DIV><SPAN class=977005513-19102010><FONT color=#0000ff size=2
face=Arial>Nikolay.</FONT></SPAN></DIV>
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style="BORDER-LEFT: #0000ff 2px solid; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px"
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<FONT size=2 face=Tahoma><B>From:</B>
freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] <B>On Behalf Of
</B>Maxim Balabaev<BR><B>Sent:</B> Monday, October 18, 2010 2:32
PM<BR><B>To:</B> freeswitch-users@lists.freeswitch.org<BR><B>Subject:</B>
[Freeswitch-users] SPA921 Problem Bad Session Description<BR></FONT><BR></DIV>
<DIV></DIV>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:
<DIV>
<DIV>
<DIV>
------------------------------------------------------------------------</DIV>
<DIV>recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:</DIV>
<DIV>
------------------------------------------------------------------------</DIV>
<DIV> INVITE sip:1001@xxx SIP/2.0</DIV>
<DIV> Via: SIP/2.0/UDP
xxx:5060;branch=z9hG4bK-ede98b83;rport</DIV>
<DIV> From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0</DIV>
<DIV> To: <sip:1001@xxx></DIV>
<DIV> Call-ID: 14d793ed-1abf9d0e@xxx</DIV>
<DIV> CSeq: 101 INVITE</DIV>
<DIV> Max-Forwards: 70</DIV>
<DIV> Contact: <sip:022@xxx:5060></DIV>
<DIV> Expires: 240</DIV>
<DIV> User-Agent: Linksys/SPA921-5.1.8</DIV>
<DIV> Content-Length: 386</DIV>
<DIV> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS,
REFER</DIV>
<DIV> Supported: replaces</DIV>
<DIV> Content-Type: application/sdp</DIV>
<DIV> </DIV>
<DIV> v=0</DIV>
<DIV> o=- 603544 603544 IN IP4 xxx</DIV>
<DIV> s=-</DIV>
<DIV> c=IN IP4 xxx</DIV>
<DIV> t=0 0</DIV>
<DIV> m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101</DIV>
<DIV> a=rtpmap:8 /8000</DIV>
<DIV> a=rtpmap:0 /8000</DIV>
<DIV> a=rtpmap:2 G726-32/8000</DIV>
<DIV> a=rtpmap:4 G723/8000</DIV>
<DIV> a=rtpmap:18 G729/8000</DIV>
<DIV> a=rtpmap:96 G726-40/8000</DIV>
<DIV> a=rtpmap:97 G726-24/8000</DIV>
<DIV> a=rtpmap:98 G726-16/8000</DIV>
<DIV> a=rtpmap:101 telephone-event/8000</DIV>
<DIV> a=fmtp:101 0-15</DIV>
<DIV> a=ptime:30</DIV>
<DIV> a=sendrecv</DIV>
<DIV>
------------------------------------------------------------------------</DIV>
<DIV>send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:</DIV>
<DIV>
------------------------------------------------------------------------</DIV>
<DIV> SIP/2.0 400 Bad Session Description</DIV>
<DIV> Via: SIP/2.0/UDP
xxx:5060;branch=z9hG4bK-ede98b83;rport=5060</DIV>
<DIV> From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0</DIV>
<DIV> To: <sip:1001@xxx>;tag=ytUD271ypvy6r</DIV>
<DIV> Call-ID: 14d793ed-1abf9d0e@xxx</DIV>
<DIV> CSeq: 101 INVITE</DIV>
<DIV> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39
2010-10-18 03-19-16 -0400</DIV>
<DIV> Accept: application/sdp</DIV>
<DIV> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
INFO, REGISTER, REFER, NOTIFY</DIV>
<DIV> Supported: timer, precondition, path, replaces</DIV>
<DIV> Allow-Events: talk, hold, refer</DIV>
<DIV> Content-Length: 0</DIV>
<DIV> </DIV></DIV></DIV></BLOCKQUOTE></BODY></HTML>