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
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