Unfortunately, the model in question (921) has latest fw.<br>For some odd reason, Linksys decided not to issue 5.2.x and 6.x fws for 9x1 models, even though phones are practically same as 9x2 counterparts (PoE is the difference).<br>
Bad luck.<br><br>Maybe you can play with codec name params in web interface of Linksys.<br><br><br><div class="gmail_quote">On Tue, Oct 19, 2010 at 4:37 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">We already told him this when he filed a JIRA on it.<br>
It's clearly broken and probably fixed with a firmware update to the device.<br>
<div><div></div><div class="h5"><br>
<br>
On Tue, Oct 19, 2010 at 9:23 AM, Kristian Kielhofner <<a href="mailto:kris@kriskinc.com">kris@kriskinc.com</a>> wrote:<br>
> Maxim,<br>
><br>
> That SDP looks nasty:<br>
><br>
> a=rtpmap:8 /8000<br>
> a=rtpmap:0 /8000<br>
><br>
> These are static payload types and don't require an rtpmap line (the<br>
> "offer" is in the m= line). However, when you use an rtpmap you must<br>
> use the IANA payload type names: PCMU and PCMA in this case. Either<br>
> of these two SDPs would be valid:<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: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>
> -or-<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 PCMA/8000<br>
> a=rtpmap:0 PCMU/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>
> Yours isn't :).<br>
><br>
> <a href="http://www.iana.org/assignments/rtp-parameters" target="_blank">http://www.iana.org/assignments/rtp-parameters</a><br>
><br>
> On Mon, Oct 18, 2010 at 6:32 AM, Maxim Balabaev <<a href="mailto:balabaev.m@gmail.com">balabaev.m@gmail.com</a>> wrote:<br>
>> I can`t make calls from linksys spa921 because of "400 Bad Session<br>
>> Description", incoming are ok. pap2t works perfectly. freeswitch is rev from<br>
>> 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<br>
>> -0400<br>
>> Accept: application/sdp<br>
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,<br>
>> REGISTER, REFER, NOTIFY<br>
>> Supported: timer, precondition, path, replaces<br>
>> Allow-Events: talk, hold, refer<br>
>> Content-Length: 0<br>
>><br>
>> _______________________________________________<br>
>> FreeSWITCH-users mailing list<br>
>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
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>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>><br>
>><br>
><br>
><br>
><br>
> --<br>
> Kristian Kielhofner<br>
> <a href="http://www.astlinux.org" target="_blank">http://www.astlinux.org</a><br>
> <a href="http://blog.krisk.org" target="_blank">http://blog.krisk.org</a><br>
> <a href="http://www.star2star.com" target="_blank">http://www.star2star.com</a><br>
> <a href="http://www.submityoursip.com" target="_blank">http://www.submityoursip.com</a><br>
> <a href="http://www.voalte.com" target="_blank">http://www.voalte.com</a><br>
><br>
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><br>
<br>
<br>
<br>
</div></div>--<br>
Anthony Minessale II<br>
<br>
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<div><div></div><div class="h5"><br>
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</div></div></blockquote></div><br>