[Freeswitch-users] SPA921 Problem Bad Session Description

[Anthony Minessale]

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 at 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 I'm not sure, but 5.1.8 firmware on linksys is quite outdated, isn't it?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 at 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 at xxx SIP/2.0
>> >    Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport
>> >    From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
>> >    To: <sip:1001 at xxx>
>> >    Call-ID: 14d793ed-1abf9d0e at xxx
>> >    CSeq: 101 INVITE
>> >    Max-Forwards: 70
>> >    Contact: <sip:022 at 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 at xxx>;tag=8f6dbe61c1d0552ao0
>> >    To: <sip:1001 at xxx>;tag=ytUD271ypvy6r
>> >    Call-ID: 14d793ed-1abf9d0e at 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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>>
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>
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-- 
Anthony Minessale II

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