On Wed, 2012-07-18 at 02:27 -0400, Jeremy Kister wrote:
I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10.
The system itself is happy and phone calls (between two parties) seem fine.
Unfortunately, when a caller listens to a Playback recording, there
seems to be
Good morning,
I have an asterisk 1.4.43 running here in the company. We are facing the
following difficulty. We have one that is responsive in a queue and logs
through agents (we still use the AgentCallBackLogin for this). Works perfectly
queue receptive.
The problem is now the need to
On 7/19/2012 3:50 AM, Hans Witvliet wrote:
Perhaps system too busy, disk not fast enough?
before doing a play-back, run iostat 1 in another window
interesting. the stutter certainly correlates to minor amounts of disk
i/o. when there is no stutter, there is nothing to report. but a minor
On 7/19/2012 10:22 AM, Muneeb Iqbal wrote:
We are pleased to announce special offer on the Singapore Numbers.
You know Singapore as a country only has a few million people, but one
of the highest per Capita Income ?
If you have not Yet started to sell them, HERE is your chance to offer
On 7/19/2012 10:50 AM, Muneeb Iqbal wrote:
My apologies Rusty
Will be careful in future.
Regards,
Muneeb Iqbal - (Ricky)
Manager Sales
Fb.com/muneebiqbal
Skype: JustMuneeb
Thanks. I forgot to mention that asterisk-biz is the appropriate place
for that type of post.
--
Rusty Newton
On Thu, Jul 19, 2012 at 11:34 AM, Eduardo eduardo.leo...@yahoo.com.br wrote:
I have an asterisk 1.4.43 running here in the company. We are facing the
following difficulty. We have one that is responsive in a queue and logs
through agents (we still use the AgentCallBackLogin for this). Works
Got it :)
Cheers
Muneeb Iqbal
On Thu, Jul 19, 2012 at 9:03 PM, Rusty Newton rnew...@digium.com wrote:
On 7/19/2012 10:50 AM, Muneeb Iqbal wrote:
My apologies Rusty
Will be careful in future.
Regards,
Muneeb Iqbal - (Ricky)
Manager Sales
Fb.com/muneebiqbal
Skype: JustMuneeb
Hi all,
I'm having trouble with the subject. I've setup the Tigase server,
registered clients and I'm seeing PubSub messages on both test servers.
Phones connected to the server with the resource I'd like to watch,
DAHDI/1, are seeing the proper states but phones on the remote systems
are
Hi list.
I have Asterisk installed on a Debian 1.8 6 64-bit.
What happens is the following, some channels are not being hangup properly.
They run the hangup in dialplan, but the output of the command core show
channels shows several channels with status rsrvd. Checking the server's
memory, the
On 07/19/2012 03:49 PM, Rodrigo Lang wrote:
I tried to shut down the channels with the command channel request
hangup Khomp_SMS/B0C2-0 (Khomp_SMS/B0C2-0 the channel is locked), but
nothing happens. I can only release these channels when I restart asterisk.
You will probably need to ask the
Hi,
Let say I have 8 PSTN line on dahdi 1~8. When a call come to dahdi 8, I
want it play a particular IVR and terminate the call. I'm looking at
elastix's GUI but seems no setting about it.
Please advice.
Thanks a lot :)
BR,
Anam.
--
You have to use from-zaptel in your context, and define your Zap DIDs in
elastix. And lastly set inbound route with your zap did defined and forward
that to your ivr.
Sent from my iPhone
On Jul 20, 2012, at 7:30 AM, Satria Anamarta anam.satri...@gmail.com wrote:
Hi,
Let say I have 8 PSTN
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