As I fairly know, you cannot send a SIP message if the telephone is not registered to you (In case host=dynamic in configuration), since you don't know which is the telephone address or if it's behind NAT.<br><br>If not registered you should try with an outside script like this: <a href="http://www.voip-info.org/wiki/view/Polycom+reboot+hardphone+script">http://www.voip-info.org/wiki/view/Polycom+reboot+hardphone+script</a><br>
<br>Regards<br clear="all">Jose Flores Galicia<br><<<a href="mailto:FloJoSe@gmail.com">FloJoSe@gmail.com</a>>><br>BriefCode && Code Based Training<br>
<br><br><div class="gmail_quote">2010/4/7 Jake Wilson <span dir="ltr"><<a href="mailto:jake.wilson@answeron.com">jake.wilson@answeron.com</a>></span><br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
We are running a PBX box for our small company's phone system. There are two phones on our network that are not working though.<br>
<br>
When I reboot a phone that works correctly, like this (note 4907 is the phones extension):<br>
<br>
asterisk -rx "sip notify polycom-check-cfg 4907"<br>
<br>
It returns this:<br>
<br>
Sending NOTIFY of type 'polycom-check-cfg' to '4907'<br>
<br>
and the phone proceeds to reboot normally.<br>
<br>
However if I try to do this with a phone that is not connecting, like extension 4912, I get this:<br>
<br>
Could not create address for '4912'<br>
<br>
And /var/log/asterisk/full just spits out:<br>
<br>
[Apr 7 15:20:41] VERBOSE[2899] logger.c: -- Remote UNIX connection<br>
[Apr 7 15:20:41] VERBOSE[12115] logger.c: -- Remote UNIX connection disconnected<br>
<br>
Not much help. I can't seem to see any other Logs that are logging anything for this phone.<br>
<br>
Does anyone have any clue where I can start to determine the problem? Where should I be looking?<br>
<br>
On this PBX box, all the Polycom provisioning configs and settings are in /home/PlcmSpIp/<br>
<br>
I'm new to the PBX/Asterisk thing cause I'm taking over for the previous IT guy and trying to figure out the mess he made with this. Any help would be appreciated.<br>
<br>
Jake<br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br>