[asterisk-dev] is it possible to retrieve if a channel has early media? (was: Dial() generating ringback until answer or early media)

[Klaus Darilion]

Hi all!

Is it somehow possible to find out if a channel is already in 
early-media state?

AFAIS there is no such flag in struct ast_channel. For implementing this 
automatic behavior in Dial() we would need such a flag.

regards
klaus

Alec Davis schrieb:
> I've updated http://bugs.digium.com/view.php?id=14504 to include option 'R'
> but I've run out of evening here in NZ.
> 
> We are using Dial option 'r(tone)' which works better for us, as we need to
> be able to signal 'Congestion' to the calling switch if Asterisk cannot
> deliver the call.
> 
> Option 'R' will indicate 'Ringing' to the calling switch, which then will
> not (in our case) failover on non delivery to the more expensive Telco
> provider.
> 
> I was unable to exhaustively test the Early Media aspect.
> 
> Alec Davis
> 
> 
> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com
> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Klaus Darilion
> Sent: Wednesday, 25 February 2009 08:58 p.m.
> To: Asterisk Developers Mailing List
> Subject: Re: [asterisk-dev] Dial() generating ringback until answer or early
> media
> 
> 
> 
> Alec Davis schrieb:
>> I posted a patch yesterday http://bugs.digium.com/view.php?id=14504 
>> that allows you to play a ringback tone from indications.conf, I 
>> hadn't seen bug 10934. Maybe I've done some of the work already 
>> required, and are able to combine what you suggest this.
> 
> Yes, that sounds good. I think a new option is needed, e.g. e (like early
> media).
> 
> e([tone]): If a tone is specified, it behaves like your current patch and
> generates a certain tone. If tone is not specified, it will generate the
> tone dynamically based on the control frames received on the outgoing
> channel. (AST_CONTROL_RINGING -> ring, AST_CONTROL_CONGESTION 
> -> congestion)
> 
> thanks
> klaus
> 
>> I need to have a good look at 10934.
>> Alec
>>
>> -----Original Message-----
>> From: asterisk-dev-bounces at lists.digium.com
>> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Klaus 
>> Darilion
>> Sent: Wednesday, 25 February 2009 09:42 a.m.
>> To: Asterisk Developers Mailing List
>> Subject: Re: [asterisk-dev] Dial() generating ringback until answer or 
>> early media
>>
>> Kevin P. Fleming wrote:
>>> Klaus Darilion wrote:
>>>
>>>> To solve this the following behavior is needed: if Dial() receives 
>>>> an AST_CONTROL_FRAME like RINGING or CONGESTION, it should check if 
>>>> the incoming channel has already early media (is in state PROGRESS?) 
>>>> and if yes, activate the tone generator with the corresponding tone, 
>>>> until PROGRESS (early media), ANSWER or HANGUP is received on the 
>>>> outgoing channel.
>>>>
>>>> There was once a patch which was related to this,
>>>> http://bugs.digium.com/view.php?id=10934
>>>>
>>>> but I think it should be solved in a more general way. (Either 
>>>> always or by adding a new Dial option).
>>> I agree that this seems like a reasonable course of action.
>> I fear my skills are yet to low to implement this. Should I add it to 
>> the bugtracker?
>>
>> klaus
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-dev mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-dev mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-dev
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
> 
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev