Marco Mouta
2007-Jan-11 04:37 UTC
[asterisk-users] Has been working for 9 Months - Very Very Strange I cannot dial specific extensions from my dialplan - NOT A CONTEXT PROBLEM!!
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@194.117.36.75:5060
To:sip:4214@XXX.XXX.XX.XX:5060
| |(2752) ------------------> (5060) |
|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
| |(2752) ------------------> (5060) |
|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
| |(2752) ------------------> (5060) |
|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
| |(2752) ------------------> (5060) |
|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
| |(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@XXX.XX.XX.XX:5060
To:sip:*98@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@XXX.XX.XX.XX:5060
To:sip:*98@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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Steven
2007-Jan-12 14:45 UTC
[asterisk-users] Re: Has been working for 9 Months - Very Very StrangeI cannot dial specific extensions from my dialplan - NOT ACONTEXT PROBLEM!!
Is there a local dialplan on the phone?
Maybe these phones were recently upgraded or reset to factory and lost the 4XXX
dialplan.
That is where I would start.
--
--
Steven
http://www.glimasoutheast.org
"Marco Mouta" <marco.mouta@gmail.com> wrote in message
news:116fd70d0701110337u79a180abpac7759ef888d1f1a@mail.gmail.com...
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@194.117.36.75:5060
To:sip:4214@XXX.XXX.XX.XX:5060
| |(2752) ------------------> (5060) |
|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
| |(2752) ------------------> (5060) |
|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
| |(2752) ------------------> (5060) |
|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
| |(2752) ------------------> (5060) |
|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
| |(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@XXX.XX.XX.XX:5060
To:sip:*98@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@XXX.XX.XX.XX:5060
To:sip:*98@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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