[asterisk-users] Has been working for 9 Months - Very Very Strange I cannot dial specific extensions from my dialplan - NOT A CONTEXT PROBLEM!!

[Marco Mouta]

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:4214 at 194.117.36.75:5060
To:sip:4214 at XXX.XXX.XX.XX:5060
|         |(2752)   ------------------>  (5060)   |
|11,721   |         INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A
GS...elephone-eve)          |SIP From: sip:4214 at 194.117.36.75:5060
To:sip:4214 at XXX.XXX.XX.XX:5060
|         |(2752)   ------------------>  (5060)   |
|12,727   |         INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A
GS...elephone-eve)          |SIP From: sip:4214 at 194.117.36.75:5060
To:sip:4214 at XXX.XXX.XX.XX:5060
|         |(2752)   ------------------>  (5060)   |
|14,739   |         INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A
GS...elephone-eve)          |SIP From: sip:4214 at 194.117.36.75:5060
To:sip:4214 at XXX.XXX.XX.XX:5060
|         |(2752)   ------------------>  (5060)   |
|18,762   |         INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A
GS...elephone-eve)          |SIP From: sip:4214 at 194.117.36.75:5060
To:sip:4214 at XXX.XXX.XX.XX: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:4214 at XXX.XX.XX.XX:5060
To:sip:*98 at XXX.XX.XX.XX: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:4214 at XXX.XX.XX.XX:5060
To:sip:*98 at XXX.XX.XX.XX: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
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