Hi all, 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.
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. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK 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! SIP to PSTN - OK SIP to IAX - OK This is a graph from ethereal: Dialing 4214, my own SIP extension! |Time | 192.168.34.26 | XXX.XXX.XX.XX | |11,219 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) ------------------> (5060) | |11,721 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) ------------------> (5060) | |12,727 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) ------------------> (5060) | |14,739 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) ------------------> (5060) | |18,762 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) ------------------> (5060) | Dialing *98 to check voicemail: 2 |21,882 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) ------------------> (5060) | 2 |21,884 | 407 Proxy Authentication Required |SIP Status | |(2752) <------------------ (61414) | 2 |21,886 | ACK | |SIP Request | |(2752) ------------------> (5060) | 2 |21,990 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) ------------------> (5060) | 2 |21,991 | 100 Trying| |SIP Status | |(2752) <------------------ (61414) | 2 |21,997 | 200 OK SDP ( g711A GSM g711U telephone-event) |SIP Status | |(2752) <------------------ (61414) | 2 |22,034 | RTP (g711U) |RTP Num packets:116 Duration:2.315s ssrc:490185229 | |(42576) ------------------> (18670) | 2 |22,208 | ACK | |SIP Request | |(2752) ------------------> (5060) | 2 |23,025 | RTP (g711U) |RTP Num packets:75 Duration:1.484s ssrc:1496378340 | |(42576) <------------------ (18670) | 2 |24,523 | BYE | |SIP Request | |(2752) ------------------> (5060) | 2 |24,525 | 200 OK | |SIP Status | |(61413) <------------------ (5060) | 2 |25,026 | BYE | |SIP Request | |(2752) ------------------> (5060) | 2 |25,027 | 200 OK | |SIP Status | |(61413) <------------------ (5060) | 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! I hope someone can point me out where is the problem! This server has only sip extensions. P4 - 1G RAM wiht TE110P with weekly reboot. Best regards, Marco Mouta
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users