[Asterisk-Users] OT: Snom 320, displaying text on the screen from *

[Sven Fischer (support)]

BTW regarding AOC you may want to have a look into this:

http://www.snom.com/wiki/index.php/Advice_of_charge_%28AOC%29_in_SIP

On Friday 10 March 2006 04:37, Dofear wrote:
> Can this feature be used to display total balance left (for a phone) on the
> display of the phone?
>
>   _____
>
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Colin
> Anderson Sent: Friday, March 10, 2006 2:33 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] OT: Snom 320, displaying text on the screen
> from *
>
>
> try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of
> registrar>"
>
> the trick is to specify the "-O desktop" parameter + the "-H <ip of
> registrar>" parameter. Sipsak fakes the host-header of the registrar so
> that the Snom thinks it is coming from your Asterisk server, then lets the
> message through to the "desktop" (the display of the phone)
>
> I wasn't kidding about obscure syntax, sipsak is a PITA
>
> -----Original Message-----
> From: Sean Kennedy [mailto:skennedy at qualitydentists.com]
> Sent: Thursday, March 09, 2006 5:01 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] OT: Snom 320, displaying text on the screen
> from *
>
>
> I have that set, but for some reason I get errors when I try sipsak, and
> nothing comes through to the phone:
>
>
>
> sipsak -M -B "test" -s sip:44 at 192.168.1.67
> timeout after 500ms
> timeout after 500ms...
>
>
> Some debugging info:
>
>
> [root at firewall root]# sipsak -vvv -M -B "test" -s sip:44 at 192.168.1.67
> warning: ignoring -i option when in usrloc mode
> fqdnhostname: 192.168.1.1
> our Via-Line: Via: SIP/2.0/UDP
> 192.168.1.1:34213;branch=z9hG4bK.105fb86e;rport;alias
>
> New message with Via-Line:
> MESSAGE sip:44 at 192.168.1.67 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.1:34213;branch=z9hG4bK.105fb86e;rport;alias
> To: sip:44 at 192.168.1.67
> Call-ID: 2089538687 at 192.168.1.1
> CSeq: 1 MESSAGE
> Content-Type: text/plain
> Max-Forwards: 70
> User-Agent: sipsak 0.9.5
> From: sip:sipsak at 192.168.1.1:34213;tag=7c8bd47f
> Content-Length: 4
>
> test
> sending message ...
>
> request:
> MESSAGE sip:44 at 192.168.1.67 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.1:34213;branch=z9hG4bK.105fb86e;rport;alias
> To: sip:44 at 192.168.1.67
> Call-ID: 2089538687 at 192.168.1.1
> CSeq: 1 MESSAGE
> Content-Type: text/plain
> Max-Forwards: 70
> User-Agent: sipsak 0.9.5
> From: sip:sipsak at 192.168.1.1:34213;tag=7c8bd47f
> Content-Length: 4
>
> test
> send to: UDP:192.168.1.67:5060
>
> ignoring MESSAGE retransmission
> timeout after 500 ms
>
>
> So I am at a bit of a loss.
>
> Thanks for your help though, I apprecaite it.  :)
>
> Colin Anderson wrote:
>
>
> Trick with Sipsak is you have to change the network port to 5060 or sipsak
> messages never hit the right port. In the web interface, Advaced > Avanced
> Network > Network identity (port): change that to 5060 and you should be
> good assuming you can figure out sipsak's nasty syntax. hth.

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