I have an Asterisk server connected to a Nortel Pbx via an E1.
Everything works fine, I get calls in and out with callerid. The
problem that has been reported to me is the following scenario:
A call comes in from the PSTN and is answered by Asterisk. The person
dials the operator (1000)
well with R6.0 and later. I have a
simular setup but with Cisco UCM but the calls come into the Nortel
first and then can be passed back and forth between them with no problem.
On 04/24/2012 10:39 AM, Carlos Chavez wrote:
I have an Asterisk server connected to a Nortel Pbx via an E1
On Tue, 2012-04-10 at 15:15 -0500, Todd Routhier wrote:
What I am trying to accomplish is to run an AGI script each time an
agent's line starts ringing. I currently have the AGI firing when the
agent answers the call using the Queue command, something like
queue(MyQueue,MyAgi.php). Works
On Mon, 2012-02-27 at 18:28 +, Noah Engelberth wrote:
I’ve been tasked with finding and implementing a CDR/Queue analyzer to
provide information to management about the call center’s performance.
My Google-fu seems to be returning a lot of things that are more or
less abandoned projects.
I am having a strange problem with an external SIP phone. It can
register and receive calls but it cannot initiate any calls. A
softphone on the same network works without problems.
As far as I can notice the difference is that the hard phone is not
sending the proper contact
Just a warning to people trying to use ooh323 with Asterisk 1.8.7. The
example config file that comes with asterisk is called chan_ooh323.conf
when it actually should be named ooh323.conf for it to work. Sent me
into a panic when I was trying to install an H323 link to an Avaya
server
On Tue, 2011-09-27 at 03:47 -0700, bilal ghayyad wrote:
Dears;
I am facing now a problem in the recording the calls that coming via the
queue, the problem that I am not able to make the filename contains the agent
(for example its extension) who received the call.
Actually by looking
Voicemail indication on the FXS port? I you have voicemail configured
the ring is indicating that the extension has a message waiting.
On Thu, 2011-09-01 at 14:51 -0500, Chris Ramirez wrote:
I have recently setup Trixbox 2.6.1 on a machine and configured it
with an FXO and FXS module.
. Is there
anyone that could/would share experiences using that?
We need to replace an old PBX interfaced to 50 FXS and 8 BRI ISDN in
Italy.
My concern is about reliability of USB
Any success stories with it? Tips and tricks?
Gilles wrote:
On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez
On Tue, 2011-08-30 at 01:31 +0200, Gilles wrote:
On Sat, 27 Aug 2011 09:31:12 -0600, linux guy linuxguy...@gmail.com
wrote:
I'm looking for an FXO device to connect to a POTS line that communicates
via USB or Ethernet.
For USB, AFAIK, there's only the one from Sangoma. All others are
We are having a problem when trying to use originate or AMI to make a
call. We have an Asterisk 1.8.5.0 server which uses a SIP provider to
call the PSTN. When dialing from IP phones everything works fine. When
you try making the call with originate, AMI or a call file then the
remote
On Tue, 2011-07-19 at 03:23 -0700, bilal ghayyad wrote:
Hi All;
I succeeded now to configure callpickup so if the SIP user pressed *8 then it
will pickup the call within the group.
What is the possibility to have another code (for example *7 or any thing
else) to pickup the call from
On Tue, 2011-07-19 at 08:53 -0700, bilal ghayyad wrote:
Hi All;
Actually what I am looking into is a method to have multiple Asterisk Boxes
to be working togethor as one entity so a distributing for the load and for
the tasks can be acheived.
I need such kind of protocols to be used in
Lately I have been getting many complains that Eyebeam crashes when you
dial a number that does not exist. This happens in both R2 and ISDN PRI
lines. The softphone stops working and has to be restarted. The
response I got from tech support was:
the actual issue is that asterisk should
On Wed, 2011-06-29 at 18:12 -0400, Alex Balashov wrote:
Perhaps do this instead?
allow=g723
allow=g729
disallow=all
On 06/29/2011 05:57 PM, Ernie Dunbar wrote:
This *should* be something that's easy to fix, but apparently I'm not
doing something right.
Our SIP long
On Fri, 2011-06-17 at 13:54 -0400, vip killa wrote:
Is there any to have asterisk record a file then send that file to
a distribution list of voicemail boxes?
What I'm trying to accomplish is a prompt for a user to record/listen
to their message and then choose to send the recording to
On Fri, 2011-06-10 at 16:31 +0530, virendra bhati wrote:
Hi List,
I don't install from yum repository. I download tar file from
asterisk.org
On Fri, Jun 10, 2011 at 3:03 PM, Alexandru Oniciuc
alexandru.onic...@trivenet.it wrote:
What do you mean?
Did you
On Thu, 2011-05-19 at 21:10 +, satish patel wrote:
How to get rid on following.. why its Invalid ?
holler*CLI queue show queue1
queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s
holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
Agent/7201 (Invalid)
On Fri, 2011-05-20 at 17:57 +, satish patel wrote:
I do have agents in agents.conf. I am not using agentlogin apps. I am
using AddQueueMember
agent = 7101,,Agent1
agent = 7102,,Agent2
From: cur...@telecomabmex.com
To: asterisk-users@lists.digium.com
Date: Fri, 20 May 2011
We are having a problem where agents are not logging off at the end of
the day and they complain about receiving calls early the next day. Is
there a simple way to automatically log off all agents (dynamic) from
all queues at a certain hour? Or do I have to parse all queues for
agents
On Thu, 2011-05-12 at 22:17 +0200, Jonas Kellens wrote:
On 05/12/2011 07:12 PM, Carlos Chavez wrote:
On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote:
Hello,
is there some way to make Asterisk light up a certain light on an
IP-phone ?
Like MWI, the message waiting
On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote:
Hello,
is there some way to make Asterisk light up a certain light on an
IP-phone ?
Like MWI, the message waiting indicator can light up if there is
voicemail.
Could this light, or even other lights (like BLF-buttons) be used to
On Sat, 2011-05-07 at 17:04 +, salaheddine elharit wrote:
Hello List,
i need to be able to record the call transferred from iax extension
to sip extension
when i call the sip extension from the IAX extension i can record the
call without any issue
but when i receive a call
On Thu, 2011-05-05 at 18:16 +0530, Asterisk Man wrote:
Hi,
I have a simple Queue(named 1) and one Member(SIP/1119) logged into
it. Now when a caller is placed into Queue and gets connected with
Member, I want to record the call. It does record the call when I use
MixMonitor() before placing
Can anyone recommend some White Papers or Success Cases that we can use
to ease the mind of a customer that has not heard much about Asterisk? All
they know is Avaya at this point.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55
On Tue, 2011-03-22 at 15:16 +0100, Lenz Emilitri wrote:
Maybe not much from the point of view of queues, but this may make
quite a difference from the point of view of monitoring your
call-center. :)
l.
2011/3/21 satish patel satish...@hotmail.com
Hey Guys,
On Wed, 2011-03-02 at 16:33 -0500, Jerry Geis wrote:
When I switched to 1.8 from 1.4 I am getting this error
pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension
(default, s, 1)
http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
This
On Fri, 2011-02-18 at 12:36 +0100, Axelle wrote:
Hi,
I'm trying to automatically have the dialplan assign an extension to a
roaming phone on my network.
I tried the following without success:
exten = 3001,1(readop),BackGround(beep)
exten = 3001,n,Read(digito,vm-youhave,3)
exten =
On Thu, 2011-02-17 at 11:13 -0600, Danny Nicholas wrote:
__
From:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jordi
Bou
Sent:
On Wed, 2011-02-09 at 14:37 -0800, Ernie Dunbar wrote:
On Wed, 9 Feb 2011, Ernie Dunbar wrote:
We have a customer who wants to forward an extension to their cell
phone, if and only if that extension is unavailable, or when the
Dial() command times out. However, should the Dial() command
Just recently I noticed that my Asterisk 1.8 server is giving the
following error at startup:
[Feb 9 17:48:56] WARNING[7968]: loader.c:387 load_dynamic_module: Error
loading module '��Է�Vi': /usr/lib/asterisk/modules/��Է�Vi.so: cannot
open shared object file: No such file or directory
On Mon, 2011-02-07 at 10:44 -0500, Daniel - Asterisk wrote:
Hi Danny,
Could you please let me know what function do I use to get if the
queue is full?
Elder
On Mon, Feb 7, 2011 at 10:42 AM, Danny Nicholas da...@debsinc.com
wrote:
I am having a problem trying to use originate from the CLI on Asterisk
1.8.2.3. The SIP peer is defined correctly and it works if I dial using
my IP phone. When I try to dial from the CLI I get this message:
pbxoficina*CLI originate SIP/protel-out/0445540881644 application playback
:-)
Background does not work inside Macros. Use READ to get the digits you need.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
--
_
-- Bandwidth
On Thu, 2011-01-27 at 14:22 -0700, Mike Diehl wrote:
Hi all,
Does anyone have any good/bad comments on the A1200P 12-port fxo/fxs
card from OpenVox?
I'll be using one to with 8-12 fxo interfaces. The cards will be
plugging into a cable-modem / phone adapter. We weren't able to port
the
On Sat, 22 Jan 2011 19:47:43 -0500, Mark Deneen wrote
On Sat, Jan 22, 2011 at 6:53 PM, Carlos Chavez
cur...@telecomabmex.com wrote:
On Sat, 22 Jan 2011 20:51:54 +, Steve Howes wrote
On 22 Jan 2011, at 18:02, Carlos Chavez wrote:
Cannot allocate memory
Have you tried looking
with the latest patches. The interval between
failures dropped from a month to a week so this is a critical concern now. I
do not s
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
On Sat, 22 Jan 2011 20:51:54 +, Steve Howes wrote
On 22 Jan 2011, at 18:02, Carlos Chavez wrote:
Cannot allocate memory
Have you tried looking at memory?
S
The server has 8gb of ram and 8gb of swap. Free indicates that there are
at least two free gb of memory and swap remains
I recently upgraded my office server to 1.8 and since then I have very
bad voice quality when calling another Asterisk server that uses 1.6.
The links is via IAX2 and I have tried using g729 and ulaw but I still
have the same problem although ulaw has a slight better result.
Any
On Wed, 2011-01-19 at 16:40 -0800, Steve Edwards wrote:
Un-top-posting...
On Wed, 19 Jan 2011, abhinav anand wrote:
I am using Asterisk version 1.6.2.5-0. Surprisingly, I don't see
dialplan reload.
If you do not have 'dialplan reload,' you do not have pbx_config.so
loaded. Since
I am having a problem trying to use originate from the CLI on Asterisk
1.8.1.1. The SIP peer is defined correctly and it works if I dial using
my IP phone. When I try to dial from the CLI I get this message:
[Jan 18 12:00:09] WARNING[3336]: chan_sip.c:19048
handle_response_invite:
with 8gb of ram. The first time
it happened we were using Asterisk 1.6.2.14 so we upgraded to 1.6.2.15 to try
and avoid the problem but no luck. Any recommendations?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
Anyone has a good recommendation for a Windows program that will open a
browser URL when your phone receives a call? We had been using Yaacid
but since it is no longer being developed we need to look for an
alternative. It should be light weight and work on all versions of
Windows.
--
Is there a way to play a different message than the periodic announce
after a certain time? I have been asked by a customer to do something
like this:
The user enters the queue.
We play position and periodic announce every 60 seconds.
If user has waited for more than 5 minutes then play
On Fri, 2010-12-24 at 07:52 +1300, CB wrote:
Could anyone recommend some documentation regarding Asterisk 1.8 and the
realtime architecture? Specifically I want to know if it is possible to set
a priority label or to use n as a priority for realtime extensions in
Asterisk 1.8? My understanding
On Wed, 2010-12-22 at 12:42 -0500, Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either
1.6.2 or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By
On Fri, 2010-12-17 at 10:40 -0500, Matt wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
Linksys Cisco SPA525 has integrated WiFi and Bluetooth
Snom 820 or 870 with optional USB adapter
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
On Thu, 2010-12-02 at 10:52 -0200, Rodrigo Lang wrote:
There is no specific Realtime database for chan_dahdi
(that I know if).
You can store the configuration using Realtime Static using
the new
chan_dahdi.conf notation without any
On Wed, 2010-12-01 at 10:15 -0200, Rodrigo Lang wrote:
Good morning list.
I wonder if I can put files and chan_dahdi dahdi_channels in real
time. Not the generals but the channels.
There is no specific Realtime database for chan_dahdi (that I know if).
You can store the
I am having a problem with a Rhino channelbank. I have an Asterisk
server running 1.6.2.9 and DAHDI 2.4.0 on a CentOS 5.5 system. We have
a TE420 card with the first port used in E1 mode (R2, 20 channels) and
the fourth is in T1 mode for the channelbank. We are using MG2 echo
I have a server running 1.6.2.13 that uses realtime for most
configurations. Everything works fine except for meetme. When I use
Meetme with Realtime any options specified in the dial plan are ignored.
For example:
exten = 1557,1,Meetme(905,icM(somemusic))
With realtime I just
On Wed, 2010-11-17 at 18:54 +, Gordon Henderson wrote:
What the score with IPv6 in Asterisk now? I've had a google about and
found the http://www.asteriskv6.org/ site but if the filename on the
download link is anything to go by its a few years old... (And a weird
anti-download
On Thu, 2010-11-11 at 18:28 -0600, Russ Meyerriecks wrote:
On 11/11/10 5:44 PM, Jeff LaCoursiere wrote:
On Thu, 11 Nov 2010, Russ Meyerriecks wrote:
On Tue, 9 Nov 2010, Daniel Tryba wrote:
I am curious about the tool dahdi_maint... what do the various
acronyms stand for?
Yea
On Thu, 11 Nov 2010 20:08:09 -0600, Russ Meyerriecks wrote
On 11/11/10 7:23 PM, Carlos Chavez wrote:
I seem to be having the same problem with a new server. I am using a
TE220 with a VPM450 module. Using Dahdi 2.4.0 and Asterisk 1.6.2.13 on
a Dell server. All calls to the outside
On Tue, 2010-11-09 at 17:38 -0800, Brett Woollum wrote:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark
5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing
[...@sipphones:1] Set(SIP/413-0005, CALLERID(num)=2) in
new
On Wed, 2010-11-10 at 11:09 -0800, Todd Fulton wrote:
Hi All,
I've got a realtime queue in place (strategy is wrandom), and have
added a member dynamically via queue add member . My agent shows in
the queue, but when he gets the call is not recognized as In Use.
Here is the output from
I just noticed that there is no Addons package for 1.8, does that mean
that I can use asterisk-addons-1.6.2.2 with Asterisk 1.8?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
signature.asc
Description: This is
On Mon, 2010-11-08 at 16:53 -0500, bakko wrote:
The addons are in the same package.
Regards
- Original Message -
From: Carlos Chavez cur...@telecomabmex.com
To: Asterisk asterisk-users@lists.digium.com
Sent: Monday, November 08, 2010 4:43 PM
Subject: [asterisk-users] Addons
On Thu, 2010-10-21 at 08:15 +0800, GBR Icasiano, Ryan A. wrote:
anyone?
regards,
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of GBR Icasiano, Ryan A.
On Fri, 2010-10-15 at 07:29 -0700, Steve Edwards wrote:
On Thu, 14 Oct 2010, bruce bruce wrote:
But it also sickens me at how badly Asterisk is made to not cope with
situations like this and worse than that is FreePBX.
Kind of like blaming the gun manufacturer instead of the criminal
I have a customer that has a Trendnet TEW-435BRM router which has the
bad habit of rewriting all external connections so the Asterisk server
only sees the IP address of the router itself. Up to today this has not
been a problem since all extensions are on the local network but now
they
On Thu, 2010-10-14 at 18:35 +0200, Daniel Tryba wrote:
On Thu, Oct 14, 2010 at 11:18:59AM -0500, Carlos Chavez wrote:
I opened up the ports on the router and my phone can register. The
problem is that I have no audio because Asterisk thinks that the phone
is on the internal network
On Mon, 2010-10-04 at 14:27 -0500, Tom Lohmuller wrote:
I am using a context to change values in a DB. Currently in my context, I
am passing it to
exten = s,1,WaitExten(7) ; 7 seconds to input
exten = s,n,Set(NEW_VAR=${EXTEN}) ;Here is my problem. This is the only
way I know how to 'grab'
then
there is no way for the PAP2T to communicate with your Asterisk server.
Maybe you could have a SIP proxy on the outside on a static IP and then allow
that Proxy to relay the PAP2T into your network?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V
the jumper settings on the card withoutopening the box
which will cause down time.
Thanks.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
On Wed, 2010-09-29 at 17:56 +0800, Songtao Yu wrote:
Hi All,
When I tried to write my dial plan as below for my FXO port, which
connects one PSTN line:
[from-pstn]
exten =s,1,Answer()
exten =s,n,Wait(1)
exten =_X.,1,Dial(DAHDI/1)
exten =_X.,n,Hangup
I got the following message:
--
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Carlos Chavez
Director de Tecnología
timing source. Since the links come from different companies
how do I choose the clock source?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
you will see
something like:
/SIP/Registry/
192.168.2.215:5060:3600::sip:x...@192.168.2.215:5060;transport=udp
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
or have people
accessing from outside..
You can put as many localnet statements as you need, one per line. I have
similar setups where up to 5 networks are internally connected so I just do
something like:
localnet=192.168.11.0/24
localnet=192.168.16.0/24
...
--
Carlos Chavez
Director de
On Tue, 2010-09-14 at 20:27 +0200, Jonas Kellens wrote:
And again !! Without me doing anything !!
PBX Core settings
-
Version: 1.6.2.11
Build Options: LOADABLE_MODULES
Maximum calls: Not set
Maximum open file handles:
On Tue, 2010-09-14 at 15:56 -0400, Zeeshan Zakaria wrote:
Hello list,
Slightly off the list topic, but I hope I'll get some help here.
Somebody wants me to implement for his project a Cisco based VoIP
system. I told him that I specialize in Asterisk based systems, but he
is not even aware
On Mon, 2010-09-13 at 12:49 +1200, Matt Riddell wrote:
On 11/09/10 12:44 PM, Carlos Chavez wrote:
The past few days I started having a problem with a small call center
setup. All agents use Eyebeam 1.5 to receive calls from a queue. Eyebeam
is
configured to auto answer the call
On Mon, 2010-09-13 at 11:22 +0100, Steve Davies wrote:
On 13 September 2010 11:07, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Gotcha. Yeah, I'm looking at implementing that (searching call
recordings by agent that took the call) here but since our asterisk call
recording is a
On Mon, 2010-09-13 at 17:48 +0200, Jonas Kellens wrote:
Hello,
can anyone please tell me how I can give arguments to my AGI script ?!
I think asterisk sees the name of the AGI + the channel as one
filename, and of course this file then does not exist.
In Asterisk 1.4 you use
not
connect...
On Mon, 2010-09-13 at 12:49 +1200, Matt Riddell wrote:
On 11/09/10 12:44 PM, Carlos Chavez wrote:
The past few days I started having a problem with a small call
center
setup. All agents use Eyebeam 1.5 to receive calls from a queue.
Eyebeam is
configured
1.4.32 just in case) on a
CentOS 5.5 x64 server with DAHDI 2.3.0.1 and a TE220B card. Could this be a
problem with chan_agent, the SIP phones or the queue? Any ideas where to
begin debugging?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55
Is there an archive of security advisories for Asterisk? We recently
upgraded a customer from 1.2 to 1.4 and now they are asking for
documentation of all security and bug related fixes. I know the
advisories get published on this list but is there an easier way to find
them than trying
I have a very difficult to diagnose problem. We are running Asterisk
1.6.2.11, DAHDI 2.4.0, FreePBX 2.8 on a Centos 5.5 server (Xeon quad
core 4gb). Last week we started having a problem where the server will
randomly stop sending and receiving calls. Asterisk does not die or
crash.
Is there any way to know if a call was transferred from reading the
CDR? Any relation in fields like UNIQUEID? Something that can be
scripted to make a special report?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext
+0x180/0x1b3
[8004c633] sys_ioctl+0x59/0x78
[8005d28d] tracesys+0xd5/0xe0
I do not know if this is normal because Wanpipe patches DAHDI or if this
indicates a problem. Any ideas or recommendations?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A
Just in case anyone is using Blockhosts
(http://www.aczoom.com/blockhosts/) with their Linux servers and
Asterisk here are the rules necessary to block invalid users:
asterisk-NoPeer:
r'Registration from .* failed for \'{HOST_IP}\' - No matching peer
found',
asterisk-NoAuth:
I have searched for some time but I have not found an asnwer on how to
fix the CDR when a call is transferred. The problem is that if someone
dials a cell phone and then transfers the call to another extensión the
CDR for the cell call stops and there is no way to track that the call
was
I am making a web interface so users can manage their voicemail. The
only problem I have is that since the Web server and Asterisk run as
different users I need to run some commands through Asterisk so I can
manipulate the voicemail files.
I know that from the CLI I can user the
You cannot use realtime static and the other realtime tables at the
same time. You will need to use realtime and then use something like
the EXEC command in sip.conf to execute a script that then pulls the
register statement from your database. Or use the realtime static table
for
On Tue, 2010-08-03 at 16:04 -0500, Danny Nicholas wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
jeremy.hellst...@synovate.com
Subject: [asterisk-users] Using SIP to dial extension that will give
anoutside line
You
I have a problem with a Sangoma card. It worked until yesterday. Now
I keep getting this error:
Jul 29 17:45:17 pbxacura kernel: wanpipe1: Enable E1 CAS signalling
mode!
Jul 29 17:45:17 pbxacura kernel: wanpipe1:w1g1: Rx Error: No
'DeviceSelect' from target: pci fatal error!
On Thu, 29 Jul 2010 20:47:58 -0500 (CDT), Tim Nelson wrote
- Carlos Chavez cur...@telecomabmex.com wrote:
I have a problem with a Sangoma card. It worked until yesterday.
Now
I keep getting this error:
Jul 29 17:45:17 pbxacura kernel: wanpipe1: Enable E1 CAS signalling
mode
On Fri, 2010-07-16 at 09:35 -0700, Kyle Kienapfel wrote:
It looks like theres no much information out there about using realtime moh
Have you tried making an extension that goes to MusicOnHold(testmoh)
On Fri, Jul 16, 2010 at 8:21 AM, Jonas Kellens jonas.kell...@telenet.be
wrote:
Hello
On Fri, 2010-07-16 at 20:06 +0200, Jonas Kellens wrote:
On 07/16/2010 07:43 PM, Carlos Chavez wrote:
Here is what I use:
CREATE TABLE `musiconhold` (
`name` varchar(80) collate utf8_unicode_ci NOT NULL,
`directory` varchar(255) collate utf8_unicode_ci NOT NULL default
On Wed, 2010-07-07 at 10:06 +0530, Hiren Mistry wrote:
Hi,
How do I configure Asterisk as a Video Conference purpose. What package
I need to configure and what steps I need to follow to configure in
dial-plan to authenticate user.
Regards,
Hiren Mistry
Asterisk is not a Video
(only for CDR) so Asterisk does not send anything configuration related to
Mysql.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
--
_
-- Bandwidth
On Mon, 2010-07-05 at 19:59 -0400, Paul Belanger wrote:
On Mon, Jul 5, 2010 at 7:29 PM, AMARDEEP SINGH yahhod...@gmail.com wrote:
Please share experience if anyone have successfully configured Dialogic
JCT-T1 card with asterisk?
Your not going to find much; there is no channel driver for
has Switchvox they were to discontinue ABE but I guess they
have not done that yet. I suppose the installed base is big enough to sustain
the product.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
On Tue, 2010-06-29 at 10:04 +0800, Zhang Shukun wrote:
hi, list
i want to know what is the best OS for install Asterisk 1.6.2.9,
which should work properly on working system.
i want to use CentOS5.2 or CentOS 5.4. Which is better and stable?
Thanks for your help.
The best
We just recently upgraded a server from Zaptel to DAHDI and Asterisk
1.4.30 to 1.6.2.9 and now we are getting this message before the server
reboots every few minutes:
Message from syslogd@ at Mon Jun 28 15:17:48 2010 ...
pbx kernel: Dazed and confused, but trying to continue
Jun 28
On Mon, 2010-06-14 at 12:00 -0400, bruce bruce wrote:
Hi Everyone,
I have a php file that if an argument is passed to it, it will echo a
number back. I am looking to use system() in dial-plan to send
${EXTEN} to it and then to get that processed value back from the php
file and put it in
On Mon, 2010-06-14 at 13:41 -0400, bruce bruce wrote:
Hi Carlso,
Thanks for the input. I have done this in php and am not familiar with
phpagi.
So, there is absolutely no way to temporarily solve this problem by
getting the value back from php file?
Wondering if it would require a
(${EXTEN})
415444555
But with the agi_extension it comes back as:
NoOp(SIP/64.111.222.111-0ca7, )
Where can I find the list of command requests that can be sent to
Asterisk? Specially that for DID.
Thanks
On Mon, Jun 14, 2010 at 2:15 PM, Carlos Chavez
cur
or not. I know I can use
grep with asterisk rx sip show peers and use that as a shell script
but I think there are better methods in Asterisk dialplan or in phpagi
that it can be check.
Thanks,
Bruce
On Mon, Jun 14, 2010 at 5:27 PM, Carlos Chavez
cur...@telecomabmex.com wrote
201 - 300 of 750 matches
Mail list logo
