On 3/4/2013 6:27 AM, Gertjan Baarda wrote:
Dear guru's
Hopefully someone can shed some light in my issue. I have created a
queue with a ringall strategy and all works fine. I want a caller to be
able to exit the queue so they can leave a message. I've added the H
option in queue command so
On 3/4/2013 7:27 AM, Gertjan Baarda wrote:
ok, resumé: When I use the n option in the queue command I can let the
caller exit the queue and send the call to a IVR-ish context and ask if
he wants to leave a message. I can timeout this an then place the call
back in the queue. When I use this
On 3/4/2013 8:00 AM, Gertjan Baarda wrote:
This will only work with the n option in the queue command and retry=0
in queue.conf. Is it not?
On Mon, Mar 4, 2013 at 2:55 PM, Alex Kauffmann akauf...@prodigy.net.mx
mailto:akauf...@prodigy.net.mx wrote:
On 3/4/2013 7:27 AM, Gertjan Baarda wrote
On 12/7/2012 6:23 AM, Vieri wrote:
Am 05.12.2012 08:48, schrieb Vieri:
Hi,
I'm trying to call out from a SIP extension to an
outbound destination via a PRI E1 (Digium B410P).
Please take a look at the PRI debug below.
# cat /etc/dahdi/system.conf
# Digium Wildcard TDM400P REV
On 12/6/2012 12:32 PM, Carlos Alvarez wrote:
We are trying to set up a system where the calls from the queue show a
specific name or number on the phone. The calls would come into one of
a few dozen DID numbers, each one for a specific company. The agent
needs to know which company the call is
34 39 3x 3x 3x 3x 3x 3x 34] added by Asterisk/DAHDI??
I've used this page as reference about frame fields:
http://www.acacia-net.com/wwwcla/protocol/q931_ie.htm
Thank you.
Giorgio Incantalupo
On 11/20/2012 05:23 PM, Alex Kauffmann wrote:
On 11/20/2012 8:03 AM, gincantalupo wrote:
Hi Leandro
On 11/20/2012 8:03 AM, gincantalupo wrote:
Hi Leandro,
I'm sure nobody has added something... tried prilocaldialplan and
pridialplan but nothing changed.
Question: if pridialplan or prilocaldialplan would work, should I see
the 0 inside PRI frame with intense debug or it is hidden?
Yes...the
On 10/25/2012 11:18 AM, Mitch Claborn wrote:
Our phone operators work off of an Asterisk queue. They take calls from
customers and take orders with our back end systems. What I need to be
able to do is tie the orders taken to the specific CDR record that
reflects the call from which the order
On 30/10/2011 05:53 a.m., Raj Mathur (राज माथुर) wrote:
On Sunday 30 Oct 2011, Sammy Govind wrote:
hmmm so IAX channel is playing with you guys.
1- Cant you guys use SIP, does this happen with SIP trunk as well !?
2- Which version of asterisk are there on both servers.
3- See the output of
Steve Totaro wrote:
Sorry for all the replies, I found the Digium PDF on Data mode.
http://www.modulo.ro/Modulo/docs/TE405-410P-user-manual.pdf
Good luck getting them to support it though ;)
I will post my Sangoma results tomorrow.
Thanks,
Steve Totaro
On Sun, Apr 6, 2008 at 10:49 AM,
Steve Totaro wrote:
Sorry for all the replies, I found the Digium PDF on Data mode.
http://www.modulo.ro/Modulo/docs/TE405-410P-user-manual.pdf
Good luck getting them to support it though ;)
I will post my Sangoma results tomorrow.
Thanks,
Steve Totaro
On Sun, Apr 6, 2008 at 10:49 AM,
Hello:
Have a TE110P laying around and decided to see if I could build a router
around it. I've tried compiling several versions of zaptel .1.4.x with
the same results. I checked the zaptel changelog and can't find
anything about it no longer being supported (or that it ever was for
that
We have several in operation but with isdn
and not R2. I know Ive seen emails from people that use them with Telmex
and have them operating, albeit with some difficulty.
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jorge Cisneros
Sent:
Claudio:
In order to receive help from this list,
you need to include more information.
How are you connecting to the carrier?
What are you using as terminals? Softphone?
Which one? SIP or IAX2? Hardphone? Brand and model.
Contents of your extensions.conf, zapata.conf,
and
We got it to work by setting the SPYGROUP
variable before every dial command for each group to be monitored and before the
call to ChanSpy by the quality agents. The way I understood the example,
both have to belong to the same SPYGROUP. So for 2
different groups, crm and sales, we use:
Setup is as follows:
Server 1
Single TE110p configured for E1
40 SIP softphone clients
Voicemail to email on 3 of the extensions
Server 2
Single TE110p configured for E1
40 IAX2 softphone clients
Voicemail to email on 2 of the extensions
Server 2 has random service interruptions with a log file
Chanspy works like a charm if all you want to do is listen to the calls.
Only problem is that it's a HEAD only feature at the moment.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Thursday, July 14, 2005 8:30 AM
To: 'Asterisk Users
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