[Freeswitch-users] incompatible destination

Sam u2nsam at gmail.com
Mon Mar 7 09:20:09 MSK 2011


Hello,


The extension 7006 was working few days ago and now it not working,rest all
the phone are working properly.


U 192.168.2.190:5060 -> 192.168.2.14:5060
  INVITE sip:7006 at 192.168.2.14:5060 SIP/2.0..Via: SIP/2.0/UDP
192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..Max-Forwards: 70..From: ""
<sip:9322273640 at 192.168.2.190>;tag=4QggZ1XcNZD3F..To: <
sip:7006 at 192.168.2.14:5060>..Call-ID: 7bf
  ae496-c324-122e-bc85-0026b976aef7..CSeq: 9403059 INVITE..Contact: <
sip:mod_sofia at 192.168.2.190:5060>..User-Agent: NOVANET..Allow: INVITE, ACK,
BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY,
PUBLISH, SUBSCRIBE.
  .Supported: timer, precondition, path, replaces..Allow-Events: talk, hold,
presence, dialog, line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer..Content-Type:
application/sdp..Content-Dis
  position: session..Content-Length: 203..X-FS-Support:
update_display..Remote-Party-ID:
<sip:9322273640 at 192.168.2.190>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH
1299446773 1299446774 IN IP4 192.168.2.190..s=FreeSWITCH
  ..c=IN IP4 192.168.2.190..t=0 0..m=audio 31474 RTP/AVP 8 101
13..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ptime:20..

U 192.168.2.14:5060 -> 192.168.2.190:5060
  SIP/2.0 488 Not Acceptable Here..Via: SIP/2.0/UDP
192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..From: <
sip:9322273640 at 192.168.2.190>;tag=4QggZ1XcNZD3F..To: "7006" <
sip:7006 at 192.168.2.14:5060>;tag=983B5A59-EB953B22..CSeq: 9403059
  INVITE..Call-ID: 7bfae496-c324-122e-bc85-0026b976aef7..Contact: <
sip:7006 at 192.168.2.14:5060>..User-Agent:
PolycomSoundPointIP-SPIP_335-UA/3.2.2.0477..Accept-Language: en..Warning:
488 SDP "NotAcceptableHere"..Content-Length: 0....



I tried all the possibilities with codec priorities on U/A .

http://pastebin.freeswitch.org/15577

Any help.

Regards
Sam
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