set this param in your sofia profile NDLB-allow-bad-iananame=true Then you can use any name you want for the name. but you can't use blank or the low level sip stack will reject it. There is no good in supporting aliases. SIP is fluid enough and the standards that actually work should be preserved. On Tue, Oct 19, 2010 at 9:26 AM, Maxim Balabaev <balabaev.m@gmail.com> wrote:
Problem was that fields with some codecs names were blank. Thats why there were "a=rtpmap:0 /8000". Also FS don`t have aliase betwen PCMA and G711.a for ex.
I'm not sure, but 5.1.8 firmware on linksys is quite outdated, isn't it? It`s really old but this is last one on their website. 2010/10/19 Brian West <brian@freeswitch.org>
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@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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