[asterisk-users] Re: Has been working for 9 Months - Very Very StrangeI cannot dial specific extensions from my dialplan - NOT ACONTEXT PROBLEM!!

[Marco Mouta]

with tcpdump  i could notice that invites didn't reach my * server.

After Rebooting Lan's Firewall CheckPoint problem solved.

On 1/12/07, Steven <asterisk at tescogroup.com> wrote:
>
>  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 at gmail.com> wrote in message
> news:116fd70d0701110337u79a180abpac7759ef888d1f1a at 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 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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