Hi all,<br><br>I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones.<br><br>Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk.
<br><br>So :<br><br>Zapata Calls to SIP extensions 4XXX - OK<br>IAX to SIP 4XXX-OK<br>SIP to SIP 4XXX - BROKEN but not for every account. Also I notice that for SIP accounts that can't dial 4XXX they can dial *98 and PSTN calls, and yes they are all in the same context since April 2006!
<br>SIP to PSTN - OK<br>SIP to IAX - OK<br><br>This is a graph from ethereal:<br><br>Dialing 4214, my own SIP extension!<br><br>|Time | <a href="http://192.168.34.26">192.168.34.26</a> | XXX.XXX.XX.XX |<br>|11,219 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From:
sip:4214@194.117.36.75:5060 To:sip:4214@XXX.XXX.XX.XX:5060<br>| |(2752) ------------------> (5060) |<br>|11,721 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From:
sip:4214@194.117.36.75:5060 To:sip:4214@XXX.XXX.XX.XX:5060<br>| |(2752) ------------------> (5060) |<br>|12,727 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From:
sip:4214@194.117.36.75:5060 To:sip:4214@XXX.XXX.XX.XX:5060<br>| |(2752) ------------------> (5060) |<br>|14,739 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From:
sip:4214@194.117.36.75:5060 To:sip:4214@XXX.XXX.XX.XX:5060<br>| |(2752) ------------------> (5060) |<br>|18,762 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From:
sip:4214@194.117.36.75:5060 To:sip:4214@XXX.XXX.XX.XX:5060<br>| |(2752) ------------------> (5060) |<br><br><br><br><br>Dialing *98 to check voicemail:<br><br>2 |21,882 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From:
sip:4214@XXX.XX.XX.XX:5060 To:sip:*98@XXX.XX.XX.XX:5060<br> | |(2752) ------------------> (5060) |<br>2 |21,884 | 407 Proxy Authentication Required |SIP Status<br> | |(2752) <------------------ (61414) |
<br>2 |21,886 | ACK | |SIP Request<br> | |(2752) ------------------> (5060) |<br>2 |21,990 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From:
sip:4214@XXX.XX.XX.XX:5060 To:sip:*98@XXX.XX.XX.XX:5060<br> | |(2752) ------------------> (5060) |<br>2 |21,991 | 100 Trying| |SIP Status<br> | |(2752) <------------------ (61414) |
<br>2 |21,997 | 200 OK SDP ( g711A GSM g711U telephone-event) |SIP Status<br> | |(2752) <------------------ (61414) |<br>2 |22,034 | RTP (g711U) |RTP Num packets:116 Duration:
2.315s ssrc:490185229<br> | |(42576) ------------------> (18670) |<br>2 |22,208 | ACK | |SIP Request<br> | |(2752) ------------------> (5060) |<br>
2 |23,025 | RTP (g711U) |RTP Num packets:75 Duration:1.484s ssrc:1496378340<br> | |(42576) <------------------ (18670) |<br>2 |24,523 | BYE | |SIP Request
<br> | |(2752) ------------------> (5060) |<br>2 |24,525 | 200 OK | |SIP Status<br> | |(61413) <------------------ (5060) |<br>2 |25,026 | BYE | |SIP Request
<br> | |(2752) ------------------> (5060) |<br>2 |25,027 | 200 OK | |SIP Status<br> | |(61413) <------------------ (5060) |<br><br>Also I notice, with SIP debug peer 4214 on * CLI , that when i dial from my sip phone 4XXX numbers, nothing seems to reach the asterisk Server!
<br><br>I hope someone can point me out where is the problem! This server has only sip extensions.<br><br>P4 - 1G RAM wiht TE110P with weekly reboot.<br><br>Best regards,<br>Marco Mouta<br><br>