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<DIV><FONT face=Arial size=2>Is there a local dialplan on the
phone?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Maybe these phones were recently upgraded or reset
to factory and lost the 4XXX dialplan.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>That is where I would start.</FONT></DIV>
<DIV><BR>-- <BR>-- <BR>Steven</DIV>
<DIV> </DIV>
<DIV><A
href="http://www.glimasoutheast.org">http://www.glimasoutheast.org</A></DIV>
<DIV> </DIV>
<DIV><BR> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Marco Mouta" <<A
href="mailto:marco.mouta@gmail.com">marco.mouta@gmail.com</A>> wrote in
message <A
href="news:116fd70d0701110337u79a180abpac7759ef888d1f1a@mail.gmail.com">news:116fd70d0701110337u79a180abpac7759ef888d1f1a@mail.gmail.com</A>...</DIV>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>
<P>
<HR>
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