[asterisk-dev] Google Voice to chan_gtalk

[Andrew Latham]

Very interesting...

I assume the Video code I saw was waiting on this issue.


~
Andrew "lathama" Latham
lathama at gmail.com

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On Wed, Oct 6, 2010 at 12:16 PM, David Vossel <dvossel at digium.com> wrote:
>
>
> ----- Original Message -----
>> From: "David Vossel" <dvossel at digium.com>
>> To: asterisk-dev at lists.digium.com
>> Sent: Tuesday, October 5, 2010 10:59:07 AM
>> Subject: Google Voice to chan_gtalk
>> Howdy!
>>
>> In Google Voice you have the option to forward incoming calls to your
>> gtalk account. Since we have a gtalk channel driver, it makes sense to
>> me that we should be able to receive incoming Google Voice calls
>> through gtalk to Asterisk. I set this scenario up and it worked,
>> except for one catch. Google Voice presents you with a prompt that
>> says "Press 1 to accept the call, press 2 to send to voicemail.". I
>> have RTP going and in both directions but I can not accept the call
>> because DTMF does not appear to be working. I press "1" but nothing
>> happens (I have the incoming call from google-voice to gtalk dialing a
>> sip phone on my desk)
>>
>> After more investigation I discovered chan_gtalk is trying to do
>> jingle signaling to perform the DTMF as opposed to RFC2833 or in band.
>> The gtalk servers reject the jingle signaling saying that the feature
>> is not implemented. After discovering that, I attempted both RFC2833
>> and in band DTMF and neither of those appeared to work either.
>>
>> So, my question is. Does anyone have any ideas of a method of sending
>> DTMF with chan_gtalk that might work? I am stumped :/
>>
>> I'm currently using the svn/asterisk/team/dvossel/gtalk_fixup branch
>> if anyone is interested in messing with this.
>>
>> David Vossel
>> Digium, Inc. | Software Developer, Open Source Software
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>> Check us out at: www.digium.com & www.asterisk.org
>> The_Boy_Wonder in #asterisk-dev
>
> For anyone who is interested, the DTMF signaling is done within the RTP stream via RFC2833.  I discovered this by performing a packet capture of an outbound call using the gmail client.  Asterisk was not sending DTMF correctly because it advertised telephony events as dynamic payload type 106, but was sending DTMF as payload type 101.  I should have a fix for this shortly.
>
> David Vossel
> Digium, Inc. | Software Developer, Open Source Software
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org
> The_Boy_Wonder in #asterisk-dev
>
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