with tcpdump i could notice that invites didn't reach my * server.<br><br>After Rebooting Lan's Firewall CheckPoint problem solved.<br><br><div><span class="gmail_quote">On 1/12/07, <b class="gmail_sendername">Steven
</b> <<a href="mailto:asterisk@tescogroup.com">asterisk@tescogroup.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff">
<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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://www.glimasoutheast.org</a></div>
<div> </div>
<div><br> </div>
<blockquote style="border-left: 2px solid rgb(0, 0, 0); padding-right: 0px; padding-left: 5px; margin-left: 5px; margin-right: 0px;">
<div>"<span id="st" name="st" class="st">Marco</span> <span id="st" name="st" class="st">Mouta</span>" <<a href="mailto:marco.mouta@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
<span id="st" name="st" class="st">marco</span>.<span id="st" name="st" class="st">mouta</span>@gmail.com</a>> wrote in
message <a href="news:116fd70d0701110337u79a180abpac7759ef888d1f1a@mail.gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">news:116fd70d0701110337u79a180abpac7759ef888d1f1a@mail.gmail.com
</a>...</div><div><span class="e" id="q_1101847851652aff_1">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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">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>
</span></div><p>
</p><hr>
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