[asterisk-users] Want web page to listen to meetme (WebRTC?)

[Steve Edwards]

I have a web page to do the usual meetme admin stuff -- mute, kick, etc.

Now, the client is asking if they can listen to the meetme -- click and 
audio comes out the computer speakers.


How can this be implemented? Is this a use case for WebRTC?

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Issue between Asterisk Queue and GSM gateway when trying to use call waiting feature

[Matthew Jordan]

On Fri, Dec 5, 2014 at 3:23 PM, Rodrigo Montiel
 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

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[asterisk-users] Faxing - Distinguish between fax and non-fax call

[Tech Support]

All;

I have a few customers that do a lot of faxing, both inbound and
outbound. Some use the Spandsp and some use the Digium FFA modules. What I
would like to do is when an outbound fax fails, determine whether the remote
end was a fax machine or a plain old phone. I would also like to determine
that in the dial plan if I could. Any insight at all with this would be
extremely helpful. 

Thanks;

John V.   

 

Tech Support

Tech Support

VoIP Business Solutions

240-215-3479 (Work/Fax)

  supp...@voipbusiness.us

 

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Re: [asterisk-users] Playing audio to bridged channels using ControlPlayBack

[Murthy Gandikota]

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-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@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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[asterisk-users] Asterisk 12 - Security Fix Only Notice

[Matthew Jordan]

Hey everyone!

This is a friendly reminder that Asterisk 12 will be entering security fix only
mode soon. As a Standard release of Asterisk, Asterisk 12 received one year of
maintenance fixes, and will receive one year of security fixes. Asterisk 12 was
first released on 2013-12-20 - the one year anniversary of which is just around
the corner! After 2014-12-20, additional releases of Asterisk 12 will no longer
be made. The final bug fix release of Asterisk 12 will therefore be 12.8.0.
Users of Asterisk 12 are encouraged to move to the next major version,
Asterisk 13, as soon as possible. Asterisk 13 is a Long Term Support (LTS) and
has maintenance support through 2018-10-24, with its full End of Life occurring
on 2019-10-24.

For more information on Asterisk versions and their supported lifetimes,
please see the following wiki page:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Thank you for your continued support of Asterisk!

-- 
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

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[asterisk-users] About voip gateway

[Leonel Florin]

Hay friends, I want to know how many simultaneous call can i do throughout
a voip gateway from the internet call to the normal telephony network,
because i want to see what implementation do i have to do multiple call
from internet to differents telephones.
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Re: [asterisk-users] About voip gateway

[Steve Edwards]

On Mon, 8 Dec 2014, Leonel Florin wrote:

Hay friends, I want to know how many simultaneous call can i do 
throughout a voip gateway from the internet call to the normal telephony 
network, because i want to see what implementation do i have to do 
multiple call from internet to differents telephones.


Please reply with a few more details of what you are planning on doing.

For example:

"I want my computer to originate 100 simultaneous calls to PSTN 
subscribers who have 'opted-in' to receive a 60 second political 
announcement.'


If all you want to do is route calls, OpenSIPS may be a better tool.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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