Yeah, that's what I was saying J good it fixed it.
BR
Gohar
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Friday, July 06, 2012 10:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hello list,
Kevin I agree with you on independent monitored entity for A leg while the
outbound leg has separate QoS measures. But after this thread I went to my
monitoring tool and saw that for some calls on the same asterisk setup I had
no RTP or RTCP while there were calls with both RTP and
Hi,
Create a context in AEL, or LUA and change the context=ael-context or
context=lua-context in sip.conf [default] section or for each sip user
decalred who needs to start call in context defined in AEL/LUA?
Regards,
Gohar
From: asterisk-users-boun...@lists.digium.com
Hey,
How've you configured your Outbound trunk ? DAHDI/1/04712527270 : What do
you've in your dahdi configuration file ! I doubt this /1 is the culprit
or else your DAHDI channel is not really working at all.
Regards,
Gohar A.
From: asterisk-users-boun...@lists.digium.com
Hi,
I think use of any Macro in queue can serve you well. Macro will be called
whenever the call is established to the agent. In that Macro check your
desired Agent and if condition matched trigger MixMonitor else do nothing.
Regards,
Gohar A.
From: asterisk-users-boun...@lists.digium.com
Dear Malvin,
I see Sam worked hard to post you the whole info about the application where
it clearly states the use of option a - Please change the configuration
line accordingly now and see if it works for you.
Best Regards,
Gohar
From: asterisk-users-boun...@lists.digium.com
Couldn't help LOL on Kyle's remarks. But it could be two users listening to
two different streams/calls. Obviously both can't share single mic on their
call(if they ever need it).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
:asterisk-conferencing-module-appkonference-16-is-now-av
ailablecatid=35:generalItemid=173
On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati virbh...@gmail.com wrote:
Hi Ahmed,
Konference is also an conferencing application.
On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed gohar.ah...@vopium.com
was looking for this :)
Please tell me how to close other socket from current sockets.
one more thing in my case it may be possible that
root 127.0.0.1 may be more then one then how to close them individually?
On Thu, Aug 25, 2011 at 5:09 PM, Gohar Ahmed gohar.ah...@vopium.com wrote:
Just realized
I'm not a php expert, but seems your php script is incomplete/ you are
sending to socket (fputs) but note receiving anything(fgets) :
See this page
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP will
help you.
From: asterisk-users-boun...@lists.digium.com
What I understood: you need to disconnect the AMI socket.
1) I want to disconnect connected manager into Asterisk. Is it possible ?
ß Close the $socket after you get the response.
What I understood: you need to maintain the socket until some button is
pressed to stop AMI
2) I want to
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gohar Ahmed
Sent: Thursday, August 25, 2011 4:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to know how many user is connected
What I
Hello All,
I'm a newbie and just started working on asterisk.I have recently installed ADM
and I want to know about ADM (Asterisk Desktop Manager) like its benchmarks
,issues or bugs, compatibility with which asterisk version e.t.c. and then any
Good web-based CRM recommendations.
thanx in
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