[Asterisk-Dev] SIP Phone reset from CLI

DUSTIN WILDES dwildes at pabbankshares.com
Tue Sep 2 08:44:58 MST 2003


No problem - I understand that completely.  But a phone will only reboot based on its sync parameters you've assigned it based on the SIP image it's running.  I don't think there's more security issues if some is able to edit your syncinfo.xml file more than being able to send an unsolicted NOTIFY to a Cisco phone.  (that can be done outside of asterisk, you know).

I know in our situation, it's easier on us to have a master reboot option or extension reboot option so I have some lower-techs do some administration & be able to implement this into a perl script.  But, in another location - it may be better to not have this option.

Regardless - I've posted the diff for anyone to use if they like or to modify it based on their needs.
Whatever you decide is fine with me either way.  :-)

See ya!




-----Original Message-----
From: John Todd [mailto:jtodd at loligo.com]
Sent: Tuesday, September 02, 2003 3:42 AM
To: asterisk-dev at lists.digium.com
Subject: Re: [Asterisk-Dev] SIP Phone reset from CLI


>On Fri, Aug 29, 2003 at 04:48:12PM -0400, James Golovich wrote:
>>  I don't know if this should really be in cvs, unless its a feature that is
>>  supported by all phones and not just the cisco phones.
>
>	I tend to agree.
>
>	But I'd like to see asterisk be able to pass the notify
>messages itself, so if I send a notify message to
>the asterisk server, i'd like it to forward them to the phones.
>
>	i have a perl script that you can use that takes
><exten> <ip> as arguments.
>
>	if you use an asterisk server as the <ip> it
>does not work..
>
>	seems like forwarding these would be the best way to do this.
>
>	- jared
>--
>Jared Mauch  | pgp key available via finger from jared at puck.nether.net
>clue++;      | http://puck.nether.net/~jared/  My statements are only mine.


This suggestion, and several others that require "proxy" 
functionality in Asterisk, have been made in the last few weeks.  I 
have said in the past that Asterisk doesn't make a good proxy, since 
it isn't really a proxy at all.  At the moment, I'll suggest again 
that is the case. (SER and Vocal are perfectly suitable, 
mostly-RFC-compliant proxies.)

While not being related to the exact circumstance at hand in this 
thread, this general concept of expanded SIP feature requests can be 
see in this request: 
http://bugs.digium.com/bug_view_page.php?bug_id=0000157

Don't get me wrong; I'd like to see some selectable "proxy" features, 
but I also would like to see that ability locked down very tightly on 
a global and/or per peer basis.  I don't want your script able to 
send SIP messages into my network if you happen to know the NAT'ed IP 
address of my phone, and the IP address of my Asterisk server.  :-)

JT
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