Unfortunately, the model in question (921) has latest fw. 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). Bad luck. Maybe you can play with codec name params in web interface of Linksys. On Tue, Oct 19, 2010 at 4:37 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
We already told him this when he filed a JIRA on it. It's clearly broken and probably fixed with a firmware update to the device.
On Tue, Oct 19, 2010 at 9:23 AM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Maxim,
That SDP looks nasty:
a=rtpmap:8 /8000 a=rtpmap:0 /8000
These are static payload types and don't require an rtpmap line (the "offer" is in the m= line). However, when you use an rtpmap you must use the IANA payload type names: PCMU and PCMA in this case. Either of these two SDPs would be valid:
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: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
-or-
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 PCMA/8000 a=rtpmap:0 PCMU/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
Yours isn't :).
http://www.iana.org/assignments/rtp-parameters
On Mon, Oct 18, 2010 at 6:32 AM, Maxim Balabaev <balabaev.m@gmail.com> 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@xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx> Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:022@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@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx>;tag=ytUD271ypvy6r Call-ID: 14d793ed-1abf9d0e@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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