asterisk users - Dec 2014 - Playing audio to bridged channels using ControlPlayBack

Hi masters,

I?m not an expert on this my friends, but I?m trying to understand which the
expected behaviour is from Asterisk side when you deal with the following
scenario:

Caller ?> GSM Gateway with SIM card A ?> Asterisk queue ?> extension
1000

GSM gateway with call waiting activated on SIM A
Queue with ?skip busy agent? disabled and ringall strategy.
SIP extension 1000 with call waiting activated, and member of Asterisk queue.

a) One Caller calls to SIM card A of GSM Gateway, call is forwarded to Asterisk
queue where SIP extension 1000 answers.
b) New Caller calls the same SIM card A of GSM gateway (it has call waiting
activated on the sim card), call is forwarded to Asterisk queue to the same
extension 1000 and a pop-up appears with the second call.
c) extension 1000 accepts it so put on hold first call, then try to pickup the
new one.

The thing is that the SIP re-invite with sendonly attribute can be seen from
extension 1000 to Asterisk queue, but this SIP invite is not being forwarded to
GSM gateway. So the GSM gateway keeps waiting for it and because it never
appears the 1st call is dropped.

Maybe you have had this issue in the past. I know that Im not an expert, but I
have been researching a lot and trying to vary configurations without clues.

The question is: Is it expected for the Asterisk queue to redirect this on-hold
message (SIP re-invite with sendonly media attribute) to the GSM gateway so it
can manage it call waiting feature on the same SIM card?

If we repeat the same scenario without queue intervention (i.e. call goes
directly to the extension) the SIP re-invite floods normally between Asterisk
and GSM gateway, so GSM gateway can decide what to do with the call.

I have no specific queue configuration, seems that queues.conf does not have any
parameter to allow? this behaviour of re-sending re-invite/on-hold messages.

Vendor from GSM gateway side is pointing that ?Asterisk js not resending on-hold
message?.

Thanks and sorry if my ignorance on this,
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I would  like to play audio--using controlplayback-- to 2 channels--agent and
caller- simultaneously. Tried meetme,confbridge,originate without success. Tried
redirecting the channels to a context, playing audio to the agent's channel
and then bridging the 2 channels. The problem with this is as soon as the bridge
is created the audio stops. I can provide the dialplan details, if anyone is
interested. Your help is appreciated.

Thanks
Murthy
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On Fri, Dec 5, 2014 at 3:23 PM, Rodrigo Montiel
<guevara2309 at yahoo.com.ar> wrote:
> Hi masters,
>
> I?m not an expert on this my friends, but I?m trying to understand which
the
> expected behaviour is from Asterisk side when you deal with the following
> scenario:
>
> Caller ?> GSM Gateway with SIM card A ?> Asterisk queue ?>
extension 1000
>
> GSM gateway with call waiting activated on SIM A
> Queue with ?skip busy agent? disabled and ringall strategy.
> SIP extension 1000 with call waiting activated, and member of Asterisk
> queue.
>
> a) One Caller calls to SIM card A of GSM Gateway, call is forwarded to
> Asterisk queue where SIP extension 1000 answers.
> b) New Caller calls the same SIM card A of GSM gateway (it has call waiting
> activated on the sim card), call is forwarded to Asterisk queue to the same
> extension 1000 and a pop-up appears with the second call.
> c) extension 1000 accepts it so put on hold first call, then try to pickup
> the new one.
>
> The thing is that the SIP re-invite with sendonly attribute can be seen
from
> extension 1000 to Asterisk queue, but this SIP invite is not being
forwarded
> to GSM gateway. So the GSM gateway keeps waiting for it and because it
never
> appears the 1st call is dropped.
>
> Maybe you have had this issue in the past. I know that Im not an expert,
but
> I have been researching a lot and trying to vary configurations without
> clues.
>
> The question is: Is it expected for the Asterisk queue to redirect this
> on-hold message (SIP re-invite with sendonly media attribute) to the GSM
> gateway so it can manage it call waiting feature on the same SIM card?
>
> If we repeat the same scenario without queue intervention (i.e. call goes
> directly to the extension) the SIP re-invite floods normally between
> Asterisk and GSM gateway, so GSM gateway can decide what to do with the
> call.
>
> I have no specific queue configuration, seems that queues.conf does not
have
> any parameter to allow  this behaviour of re-sending re-invite/on-hold
> messages.
>
> Vendor from GSM gateway side is pointing that ?Asterisk js not resending
> on-hold message?.
>
Asterisk is a back to back user agent. As such, it does not "forward"
or proxy any SIP messages. The re-INVITE sent from the SIP device
represented by extension 1000 in your scenario is handled by Asterisk,
and causes the channel on the other side of the bridge with that SIP
channel to be put on Hold.

There is no mechanism in Asterisk today to pass through a re-INVITE to
initiate a remote hold.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

There is one more thing to try:

http://snapvoip.blogspot.com/2009/07/appkonference-asterikast-high.html

I would appreciate if anyone can comment on the feasibility of playing an audio
file to the caller and callee using ControlPlayBack and appkonference. Much of
the reviews indicate that appkonference is an over-kill for an audio as its main
functionality is with video. Going past that.

Thanks

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Murthy Gandikota
Sent: Saturday, December 06, 2014 8:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Playing audio to bridged channels

I would  like to play audio--using controlplayback-- to 2 channels--agent and
caller- simultaneously. Tried meetme,confbridge,originate without success. Tried
redirecting the channels to a context, playing audio to the agent's channel
and then bridging the 2 channels. The problem with this is as soon as the bridge
is created the audio stops. I can provide the dialplan details, if anyone is
interested. Your help is appreciated.

Thanks
Murthy
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El Lunes, 8 de diciembre, 2014 12:51:42, Matthew Jordan <mjordan at
digium.com> escribi?:
   

 On Fri, Dec 5, 2014 at 3:23 PM, Rodrigo Montiel
<guevara2309 at yahoo.com.ar> wrote:
> Hi masters,
>
> I?m not an expert on this my friends, but I?m trying to understand which
the
> expected behaviour is from Asterisk side when you deal with the following
> scenario:
>
> Caller ?> GSM Gateway with SIM card A ?> Asterisk queue ?>
extension 1000
>
> GSM gateway with call waiting activated on SIM A
> Queue with ?skip busy agent? disabled and ringall strategy.
> SIP extension 1000 with call waiting activated, and member of Asterisk
> queue.
>
> a) One Caller calls to SIM card A of GSM Gateway, call is forwarded to
> Asterisk queue where SIP extension 1000 answers.
> b) New Caller calls the same SIM card A of GSM gateway (it has call waiting
> activated on the sim card), call is forwarded to Asterisk queue to the same
> extension 1000 and a pop-up appears with the second call.
> c) extension 1000 accepts it so put on hold first call, then try to pickup
> the new one.
>
> The thing is that the SIP re-invite with sendonly attribute can be seen
from
> extension 1000 to Asterisk queue, but this SIP invite is not being
forwarded
> to GSM gateway. So the GSM gateway keeps waiting for it and because it
never
> appears the 1st call is dropped.
>
> Maybe you have had this issue in the past. I know that Im not an expert,
but
> I have been researching a lot and trying to vary configurations without
> clues.
>
> The question is: Is it expected for the Asterisk queue to redirect this
> on-hold message (SIP re-invite with sendonly media attribute) to the GSM
> gateway so it can manage it call waiting feature on the same SIM card?
>
> If we repeat the same scenario without queue intervention (i.e. call goes
> directly to the extension) the SIP re-invite floods normally between
> Asterisk and GSM gateway, so GSM gateway can decide what to do with the
> call.
>
> I have no specific queue configuration, seems that queues.conf does not
have
> any parameter to allow? this behaviour of re-sending re-invite/on-hold
> messages.
>
> Vendor from GSM gateway side is pointing that ?Asterisk js not resending
> on-hold message?.
>
Asterisk is a back to back user agent. As such, it does not "forward"
or proxy any SIP messages. The re-INVITE sent from the SIP device
represented by extension 1000 in your scenario is handled by Asterisk,
and causes the channel on the other side of the bridge with that SIP
channel to be put on Hold.

There is no mechanism in Asterisk today to pass through a re-INVITE to
initiate a remote hold.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
Thank you master!!

   
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