[Asterisk-Users] Re: Snom190 intercom

[Shaun Dwyer]

Hi Josh, List,

I've managed to get the intercom working with the patch as available from:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom
(labelled as: For those who want a patch that doesn't affect VXML_URL, 
get it here <http://www.frogstorm.info/asterisk/snom-intercom.patch>.),
then as per the instructions from the wiki, SetVar(_INTERCOM=true) 
before dial.

This has the effect of doing what Josh said needs to be done in that 
intercom=true
is put on the end of the request URI instead of appended to the To: 
header as per what
SetVar(_VXML_URL=intercom=true) would do.

So now this is confirmed as working properly with the Snom190.

You must also enable the intercom either thru a subscribed config that 
tells the phone:
intercom_enabled&: on

or by setting it in the advanced options.

I am running firmware snom190-SIP 3.57v.

If anyone wants more detail on how I went about getting this working, 
please do email me :)

Cheers,
-Shaun

Shaun Dwyer wrote:

> Hi Josh,
>
> Thanks for the info..
>
> how did you get intercom=true into the URI, and onto the end of the 
> INVITE line?
>
> btw, I got an intresting response from Sven of Snom...
>
> [Sven Fischer (support) wrote:]
>
>> Hi,
>>
>> as far as I understood intercom will only work if you are not using 
>> any password for registering at the registrar at the moment.
>>
>> But we will add a line based auto answer functionality which should 
>> enable intercom for our phones more easily.
>>
>> regards,
>>
>> Sven Fischer
>
>
>
>
>
> Cheers,
> -Shaun
>
> Josh Dady wrote:
>
>>> As you can see from the SIP trace below (from the called phone), 
>>> intercom=true is being appended to the To: header as per requirements.
>>
>>
>>
>> The "intercom=true" needs to be appended to the request URI, not to 
>> the header as a whole -- your To: header should be:
>>
>>     To: <sip:1011 at 192.168.10.150:2051;line=9avrmhew;intercom=true>
>>
>> Mind you, I didn't get the phone to respond to the intercom=true 
>> until I added it on the request line as well, so the INVITE line of 
>> your request would be:
>>
>>     INVITE sip:1011 at 
>> 192.168.10.150:2051;line=9avrmhew;intercom=true SIP/2.0
>>
>> I'm on a Snom 220 with SIP 3.56t, and I'm stuck on the very next step 
>> of the process -- answering the phone's challenge to the INVITE 
>> request.  The wiki indicates that the Snom needs to challenge with 
>> realm=snom, but even if I add snom into our internal DNS so that I 
>> can set the registrar to snom (that being the only way I can see to 
>> change what the phone uses as realm), it still rejects the digest 
>> response.  Anyone have this working with recent loads of SIP that can 
>> shed any light on this?
>>
>>> I've email'd snom a few days ago but have yet to get a response.
>>
>>
>>
>> According to their web page, they have a new office as of April 1, 
>> and I got a response to a support request (on this very issue) today 
>> saying that they'd likely not be able to respond until people are 
>> settled into the new offices, so you'll likely have to be patient 
>> with them.
>>
>> -- 
>> Joshua P. Dady
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>