[Asterisk-Users] Snom 360 record button?

[dbruce]

If you ar waiting for the Snom to send DTMF.... you'll be waiting a VERY
long time....

The Snom sends an INFO message... An info message is not exclusively for
DTMF.... The key header here is the "Record: on" header... I would expect
that, once supported, the snom would send another INFO message to turn off
recording with a "Record: off" header.

To support this one touch record method on the Snom you will need to modify
the receive_info function to check for the "Record" header. Once it is
determined that the Record header has been receive then you would queue a
frame to transmit the DTMF for your automon function.

Regards,
Derek


----- Original Message -----
From: "Patrick Friedel" <pfriedel at copweb.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Thursday, July 28, 2005 1:37 PM
Subject: Re: [Asterisk-Users] Snom 360 record button?


> Maik Schmitt wrote:
>
> >>Is this during a call? Can you please send me a full SIP DEBUG of the
call?
> >>
> >>Brainstorming, maybe we could treat this as a transfer to a local
> >>extension somehow and turn monitor on in the dial plan that way...
> >>
> >>
> >
> >Hmm IMO the automon-feature would be better for this. It does exactly
what we
> >want (start recording during a call) and is configurable via
Dial-Options.
> >The only thing I don't know is how to activate it without sending the
DTMF
> >sequence.
> >
> >
>   Hmm, you're right, I wasn't aware of the automon feature - I don't
> know if it was in the original SIP trace, but the 360 is sending _some_
> DTMF signalling, but I don't know what it's actually sending.  I'm
> currently at 9 for verbosity, let me amp that up and see if it will ever
> actually display the tones it's receiving.
>
>
> voip*CLI> sip debug peer pjf
> SIP Debugging Enabled for IP: 10.0.1.213:2051
> voip*CLI> set verbose 255
> Verbosity was 9 and is now 255
> voip*CLI>
> [boring build up that isn't new to anyone]
>
> Sip read:
> INFO sip:95551212 at 10.0.1.11 SIP/2.0
> Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-aytv2p1rs5ih;rport
> From: "Patrick" <sip:pjf at voip>;tag=4rbnk4yyxd
> To: <sip:95551212 at voip;user=phone>;tag=as3b19835c
> Call-ID: 3c3382731d4c-bti01myl15cz at snom360
> CSeq: 3 INFO
> Max-Forwards: 70
> Contact: <sip:pjf at 10.0.1.213:2051;line=t2jii7ty>
> User-Agent: snom360/3.60r
> Record: on
> Content-Length: 0
>
>
> 11 headers, 0 lines
> Receiving DTMF!
> Jul 28 14:12:15 WARNING[26025]: chan_sip.c:6166 receive_info: Unable to
> parse INFO message from 3c3382731d4c-bti01myl15cz at snom360. Content
> Transmitting (no NAT):
> SIP/2.0 415 Unsupported media type
> Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-aytv2p1rs5ih
> From: "Patrick" <sip:pjf at voip>;tag=4rbnk4yyxd
> To: <sip:95551212 at voip;user=phone>;tag=as3b19835c
> Call-ID: 3c3382731d4c-bti01myl15cz at snom360
> CSeq: 3 INFO
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:95551212 at 10.0.1.11>
> Content-Length: 0
>
> [teardown]
>
> Hmm.  It's getting _something_, but the verbosity won't reveal it.
> Checking chan_sip.c, I don't see a mechanism for revealing the data,
> it'd be around line 7716.  It we could reveal it (or Nils just tells us.
> :), it might be as simple as editing features.conf and setting automon
> to whatever DTMF the Snom sends when you hit the record button.
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users