I am having problems with Eyebeam when the user dials a number that is
not available. This problem exists with both Asterisk 1.4 and 1.6 using
Eyebeam or Xlite. The problem seems to be that when the soft phone
receives the 503 Unavailable response it will not be able to dial
another number
On Fri, 2010-06-04 at 13:38 -0700, Sascha Ferley wrote:
Hi,
I am starting to notice some weird errors with DAHDI.
dahdi: Disabled echo canceller NLP because of CED tx detected on channel 2
...
Anyone ever see this? How would I track it down?
The version we are running is DAHDI 2.2.1,
On Fri, 2010-05-28 at 10:43 +0200, Vincent wrote:
Hello
I was editing files manually, and noticed that if I include the
familiar fxsks=1 in /etc/dahdi/system.conf, Dahdi fails loading:
=
# cat /etc/dahdi/system.conf
loadzone= fr
defaultzone = fr
fxsks=1
On Mon, 2010-05-24 at 16:56 -0400, Michael Munger wrote:
I am trying to get a zaptel install converted to dahdi.
I can get dahdi installed, and the pseudo device even shows up; however,
dahdi show channels shows me nothing. There is a TE122 and a TDM800 in there,
and neither show up.
Does anyone know if you can use softphones on thin clients? I have a
new customer that wants to use Eyebeam (about 10 users) on a thin client
platform. Each user has a little box on their desk that has a USB port,
mic and headphone jacks and monitor.
I am worried about conflicts
clients...
On 20 May 2010, at 18:35, Carlos Chavez wrote:
I am worried about conflicts when running 10 softphones on the same
server since they will all try to use por 5060.
And the fact most terminal services servers/clients still don't support
audio input.. only output
I want to make a web interface so my users can listen/erase voicemails.
Is there a way to do this from the Asterisk manager interface? Since
Asterisk and the web server do not run as the same user I cannot do a
direct manipulation of the voicemail files
in /var/spool/asterisk/voicemail.
On Tue, 2010-04-20 at 21:07 -0400, Leif Madsen wrote:
You could set an inherited channel variable as the first thing you do before
calling MixMonitor(). Something like:
exten = s,1,Verbose(2,Starting Call Recording) ; I always start my first
priority with something innocuous
exten =
On Wed, 2010-04-21 at 19:36 -0400, bruce bruce wrote:
Here are result of dahdi_test:
[r...@ip-10-251-123-3 ~]# dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434%
-434.763% 99.239% 93.770% 99.141% 99.822% 91.232%
I have a customer that needs to record all calls coming in and out.
The problem I am having is when a call comes in to the operator and it
is transferred to another extension. The first mixmonitor begins
recording when the operator picks up but the recording stops when the
call is
On Tue, 2010-04-20 at 15:04 -0400, Leif Madsen wrote:
Carlos Chavez wrote:
I have a customer that needs to record all calls coming in and out.
The problem I am having is when a call comes in to the operator and it
is transferred to another extension. The first mixmonitor begins
On Mon, 2010-04-19 at 11:19 -0500, Alyed wrote:
If that's the case what I usually do is just stop asterisk, delete the
contents of /usr/lib/asterisk/modules/ (back it up first!) and compile
the new version (don't run make samples if you want to preserve your
old .conf files).
When using
On Fri, 2010-04-16 at 09:00 -0500, Jason Walker wrote:
I know that this is a “feature” but I would like to have the hold
music recorded while a person is on hold. So I know the agent put
them on hold and not just muted.
I have
monitor-join=yes
monitor-format=wav
in my queues.conf
On Fri, 2010-04-16 at 08:38 -0700, Luki wrote:
Please note: A Zaptel timer must be present for conferencing to work!, but
if the user does not have ZAP/DAHDI hardware, he can use ZAP/DAHDI DUMMY
Actually, my understanding is that this is incorrect. The conference
must contain ZAP/DAHDI
I have a problem that when one of my SIP providers has a problem the
rest of my SIP extensions and trunks stop working until either the SIP
provider fixes the problem or Asterisk stops trying to register to that
provider. Why does this happen? A single provider having problems
should not
On Thu, 2010-03-25 at 17:11 +0100, Christian Victor wrote:
Hi James,
we did sucessfully run two TE410P with 8xE1 in a decent Intel P4-3200
machine with quite heavy line usage. No codec conversion course.
I don't believe that there is a hard limit of E1s coded into Asterisk.
But the
On Mon, 2010-03-22 at 23:40 +, ayodele abejide wrote:
I am a newbie to asterisk, I have a complete installation of asterisk
running on my ubuntu machine and I have x-lite installed also, I would
like to know if I can call myself on the same machine, because
whenever I try to call myself I
fill up
and the agents do not receive any more calls until we restart Asterisk. I
think I have traced the problem to func_devstate in those versions. When I
downgrade to 1.4.27 or lower I do not have this problem.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A
The las time I had this problem it was because I had a typo in an
exclude statement in my dialplan, all the dialplan commands were gone
from the CLI except dialplan show. Go over your dialplan and maybe
start with a new extensions.conf and start pasting parts of your
dialplan until it
worried but I would like to make sure this is not a sign of something more
serious.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
--
_
-- Bandwidth
On Wed, 2010-03-10 at 15:48 -0700, James Noble wrote:
This is off topic but I have inherited a Nortel Option 11c and am
curious what features you are using that make you want to continue
using it in conjunction with Asterisk as opposed to moving completely
over to Asterisk. I am just learning
On Thu, 2010-03-11 at 13:27 +1300, Alec Davis wrote:
The only thing they will really miss is the MWI light on their phones.
Asterisk supports VMWI Visual MWI in various forms;
From a cheap analog phone point of view:
have a look at 'mwisendtype' in chan_dahdi.conf
line
Does anyone know if Asterisk can function as a voicemail system for a
Nortel Option 11 PBX? We will be connecting Asterisk to act as an IVR
before sending calls to the Nortel and as a Voicemail system in case the
user does not answer. That part is trivial, the only problem we have is
On Thu, 2010-03-04 at 23:27 +, Steve Howes wrote:
On 4 Mar 2010, at 23:11, Steve Edwards wrote:
On Thu, 4 Mar 2010, Steve Edwards wrote:
On Fri, 5 Mar 2010, David @ULC wrote:
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create
I am developing a few AGI programs using PHPAGI. This is the first
time developing for Asterisk 1.6 and I am having a lot of problems
reading variables with the $agi-get_variable construct. While an AGI
debug shows me that I am asking for the variable and Asterisk is giving
me the
On Wed, 2010-03-03 at 10:14 +0530, DHAVAL INDRODIYA wrote:
Hi,
Carlos
I checked dmesg on my server and i found following message
what is meaning for this ? i cant understand
VPM400: Not Present
VPM450: echo cancellation for 128 channels
VPM450: hardware DTMF disabled.
VPM450:
On Tue, 2010-03-02 at 15:44 +0530, DHAVAL INDRODIYA wrote:
Dear All,
How can we know the On board supports echo cancellation
I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev
02) board
all working fine but sometimes i got echo when user are calling a PRI.
is
On Tue, 2010-03-02 at 15:59 +0100, jonas kellens wrote:
On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote:
If you are changing RealTime config in your DB you need to do a sip
prune realtime either directly from asterisk cli or using AMI. You
really do not need to do a SIP reload when
I have always heard that the less cards in a single system the better.
Why not try two Sangoma A108DE cards (8 ports each). Also make sure you
have hardware echo cancellation on the cards for this number of ports.
On Mon, 2010-02-22 at 14:20 +0100, Arjan Kroon | Mobillion wrote:
Hi,
I have a strange callerid problem. All my SIP phones display the
correct name of the caller but the number is always the number of the
extension that was called. If I do a NoOp on the dialplan I can see
that both name and number are correct.
The call log in my phone records all
On Thu, 2010-02-04 at 09:55 +, Andrew Thomas wrote:
Hello all,
Just wondering if anyone ever solved the Aastra 50-BLF limit when used
with Asterisk (any flavour)?
I know it's not strictly and Asterisk question - but I'm sure there's
plenty of you out there using Aastra's on the end.
On Tue, 2010-02-02 at 21:20 +0200, Emre Kurnaz wrote:
Hi all,
Does asterisk cache realtime sip md5secret values?
I create a user over a web site and set a password as asd and I can login
with that password. After a while I change my password and set it as 123.
Although the password is
I have a strange problem with CallerID that only affects some phones.
The problem is that whenever I receive a call the Callerid Name is
correct but the Callerid number is always my own extension. It does not
matter if the call is internal or external. So far only Aastra phones
and
On Wed, 2010-01-27 at 10:27 +0800, Zhang Shukun wrote:
hi,all
i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except
realtime queue.
it seems queue_table works fine, but queue_member_queue not work, the
two tables works fine when in 1.4.28.
is that something changed
You must read the upgrade instructions. The database definitions in
res_mysql.conf have changed. The way you reference the database in
extconfig.conf is also different.
On Mon, 2010-01-25 at 09:33 +, Ishfaq Malik wrote:
What happens when you try the command
mysql -uroot -proot
On Fri, 2010-01-22 at 20:25 +0530, Deep D wrote:
I want to do something like this
exten = 1234,n,Queue(6000,c)
exten = 1234,n,Set(CDR(userfield)=${Agent}) ;; where Agent is the
agent who answered the call
exten = 1234,n,Hangup
Actually because the user will hangup within the Queue
On Tue, 2010-01-12 at 18:05 -0500, C F wrote:
Anyone on the list ever used it?
I'm trying to quote a system with 192 analog ports, one of the options
are the Xorcom 32 channel FXS USB Channel Banks.
Any input would be appreciated.
I have used Astribanks for a while now and they are
Hi. Does anyone have a patch or workaround for the 50 BLF limit of
Aastra phones? I have a couple 57i with the 560M console and only the
first 50 BLF lines get registered. I am using the latest firmware from
Aastra but I read that this limit was imposed because of a memory leak.
On Mon, 2009-12-21 at 12:48 +, Julian Lyndon-Smith wrote:
We have a couple dozen Aastra 9133i phones in use - no problems
encountered, they worked well for us. However, these are now
discontinued. Does anyone have any views on the new product line up ,
or the Yealink phones ?
Julian
Actually it should be:
exten = _6XXX,1,Dial(SIP/${EXTEN})
What I actually do is set up all extensions in a different context and
then include that context in the IVR context. The problem with the
above line is that if extension 6456 does not exist there is still a
match.
On
On Tue, 2009-12-08 at 19:04 -0500, Mike wrote:
Thanks Tim and Danny. It seems a more direct way should be there, but
that`ll work.
A more direct way would be to use SNMP in Asterisk and keep statistics
with Cacti. That way you will have an historical view of usage by hour,
day,
on Aastra phones?
Any guidance will be highly appreciated.
--
Zeeshan A Zakaria
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
___
-- Bandwidth and Colocation Provided
is that when I try to
call out it never works. The channel hangs up immediately. I cannot even use
the originate CLI command to call out.
Anyone has experience with these adapters?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
On Tue, 2009-12-01 at 15:32 +, Joao Gomes Pereira wrote:
Hello
I'm trying to register an Asterisk working behind Nat.
Here is the trunk:
register=username:passw...@sip.startel.pt
[startel]
type=peer
host=sip.startel.pt
username=username
fromuser=username
secret=password
On Thu, 2009-11-12 at 14:50 +1100, Michael Wyres wrote:
Have you tried nat=yes in the definition in sip.conf?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Thursday, 12 November
On Tue, 2009-11-03 at 16:09 +, Lee Archer wrote:
Hi, is userfield the only extra CDR field that can be added or can
others?
Regards
Lee
With Asterisk 1.6.X there is adaptive CDR where you can add as many
fields as you want to the CDR, even using a database through ODBC or
I have been getting the following message every time I make a call for
the past few months:
[Nov 2 13:08:18] WARNING[9859]: file.c:1273 waitstream_core: Unexpected
control subclass '-1'
Everything seems to be working so I do not know if this is important.
I am using Asterisk
On Fri, 2009-10-30 at 08:37 -0500, Tilghman Lesher wrote:
On Thursday 29 October 2009 12:32:48 Carlos Chavez wrote:
On Thu, 2009-10-29 at 12:23 -0500, Tilghman Lesher wrote:
On Thursday 29 October 2009 11:49:30 Carlos Chavez wrote:
On Wed, 2009-10-28 at 23:36 -0500, Tilghman Lesher wrote
On Wed, 2009-10-28 at 23:36 -0500, Tilghman Lesher wrote:
On Wednesday 28 October 2009 17:57:49 Carlos Chavez wrote:
I am having a problem with Asterisk 1.6.2.0-rc3 and Asterisk-Addons
1.6.2.0-rc1 when recording CDR to a Mysql database. All fields except
callerid are recorded properly
On Thu, 2009-10-29 at 12:23 -0500, Tilghman Lesher wrote:
On Thursday 29 October 2009 11:49:30 Carlos Chavez wrote:
On Wed, 2009-10-28 at 23:36 -0500, Tilghman Lesher wrote:
On Wednesday 28 October 2009 17:57:49 Carlos Chavez wrote:
I am having a problem with Asterisk 1.6.2.0-rc3
I am having a problem with Asterisk 1.6.2.0-rc3 and Asterisk-Addons
1.6.2.0-rc1 when recording CDR to a Mysql database. All fields except
callerid are recorded properly after every call. I have both a clid
and callerid field in the database but both fields are empty. In
cdr_mysql.conf I
On Fri, 2009-10-23 at 22:24 +0100, Magnus Kelly wrote:
Hi all,
I'm struggling with figuring out how to get management information with
regard to where users are within a IVR system. Does anyone have any tips
on reporting process available on where users are if call to IVR is
disconnected or
On Tue, 2009-10-20 at 15:39 -0600, Joseph wrote:
On 10/20/09 09:01, Jonathan Thurman wrote:
Not likely. Cisco works great with CallManager, but seems to be
somewhat broken with anything else... wonder why? If you want
something that is dependable and easy to configure I have had great
On Mon, 2009-10-12 at 05:24 -0700, Robor Oghene wrote:
Dear All,
Can I mix realtime conf and static configuration files?
Thanks for responses.
The rule is that you can use Realtime and the text config file at the
same time. If you use Realtime Static you cannot use the text file
Starting with Asterisk 1.2 I have always used realtime static to load
my extensions.conf into Asterisk. It worked perfectly up to version
1.6.0.X but starting from 1.6.1.X and upwards it simply does nothing. I
can see that the extensions.conf file is mapped to the database:
== Parsing
On Thu, 2009-09-10 at 03:29 +0530, David @ULC wrote:
Below link show the download link for Call Aanalyzer and install
procedure :
http://www.757.org/~joat/wiki/index.php/Viewing_CDR_Data_with_Asterisk_CDR_Analyzer
But how to create DB in mysql and what wld be he structure ?
On Tue, 2009-09-08 at 13:03 +1200, Matt Riddell wrote:
On 8/09/09 5:35 AM, Carlos Chavez wrote:
Today is a strange day. My asterisk server is suddenly saying that all
extensions are on hold. All my hints are like this:
-= Registered Asterisk Dial Plan Hints
I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static
configuration for extensions.conf will not load. All other realtime
configs work (SIP, IAX2, Voicemail). I cannot find any reference or
documentation about the structure of the realtime static database for
1.6.1.x but I
On Tue, 2009-09-08 at 11:54 -0500, Miguel Molina wrote:
Carlos Chavez escribió:
I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static
configuration for extensions.conf will not load.
Just curious, is there any specific reason for you to upgrade from the
latest 1.6.0.14
I am having a strange problem today. When I arrived into the office
today two Aastra phones were not working. They can receive calls but
not make them. The models are 480i and 480i CT. Other Aastra phones
like 55i and 57i work fine, it only seems to affect the older phones.
Today is a strange day. My asterisk server is suddenly saying that all
extensions are on hold. All my hints are like this:
-= Registered Asterisk Dial Plan Hints =-
4...@hints : SIP/4101
State:HoldWatchers 0
4...@hints
(eg: outside line)??
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Monday, September 07, 2009 1:12 PM
To: Asterisk Users List
Subject: [asterisk-users] Aastra phones and Asterisk
It seems that older Aastra phones (9112i, 9133i, 480i, 480i CT) have a
problem with the new SIP implementation in Asterisk 1.6.X that makes
them unable to dial. They can receive calls but when you attempt to
dial the phone remains silent. You can see in core show channels that
the first
They are the oldest (4 years) VoIP provider here in Mexico. I have
many lines with them for my company an clients and most of the time it
works very well.
On Fri, 2009-07-17 at 07:26 +0200, Michiel van Baak wrote:
On 11:39, Thu 16 Jul 09, Carlos Chavez wrote:
Try http
I did not catch all the messages on this thread but why not use the
messages-expire.pl script included in Asterisk for this simple task? It
will delete and renumber all messages and you can program how many days
before a message is deleted.
--
Telecomunicaciones Abiertas de
Try http://www.inext.com.mx they can provide DIDs in several cities in
Mexico.
On Thu, 2009-07-16 at 09:16 +0200, Michiel van Baak wrote:
Hey all,
I was wondering if anyone knows about a Mexican ITSP I can connect to to
route calls from and to my * boxen.
If it matters: I'm
On Tue, 2009-06-30 at 16:17 -0400, Jeremy Winder wrote:
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is the inability to see
who is on the phone. Can
On Mon, 2009-06-22 at 08:51 +0200, Andrew Nowrot wrote:
Hi
I am having a problem with extension matching in asterisk (I am using
asterisk 1.6.0.6). Is there a difference between extensions matching
in realtime architecture and extensions matching in extensions.conf
file.
For example when
Is there a README file with the new Realtime SQL table definition for
Meetme?
On Mon, 2009-06-01 at 13:45 -0500, Tilghman Lesher wrote:
On Monday 01 June 2009 04:43:43 ik wrote:
I have MeetMe rooms generated dynamically and it always have two people
inside that are entered by
I have been trying out several stun servers with Asterisk 1.6.0.9 and
1.6.1.0 and I keep getting the following message:
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed
[May
Today I tried to upgrade from 1.6.0.9 to 1.6.1.0 but after installing I
get the following error over and over on the CLI:
[May 18 16:04:00] WARNING[11261]: res_config_mysql.c:317 realtime_mysql:
MySQL RealTime: Invalid database specified: asteriskcdrdb
[May 18 16:04:00] WARNING[11261]:
?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
Agentcallbacklogin was deprecated in Asterisk 1.4 and eliminated from
1.6 so you now need to use Dynamic Agents. Although they claim that is
is simple enough to replace that functionality with dial plan code I
have yet to see a one line example that replaces everything the
The bug was probably found and corrected a couple years ago so why
waste time trying to chase it down again?
On Wed, 2009-04-22 at 20:59 +0200, alemo...@legos.fr wrote:
Thanks for your answer Steve.
So if I understand correctly, the best solution is to update Asterisk
rather than
Last time I upgraded Zaptel to DAHDI I had a similar problem until I
erased the zaptel modules. The problem is that the Zaptel modules load
before DAHDI and you have a conflict with Asterisk. Delete everything
from /lib/modules/(Kernel Version)/misc and then reinstall DAHDI.
On Fri,
I just upgraded a very old Asterisk installation to the last 1.2.31 I
can find in Asterisk.org site. Now for some reason my IAX clients
cannot connect to the server. I can do a iax2 show peer iaxmodem1 and
I get this:
* Name : iaxmodem1
Secret : Set
Context :
No, it is the same one. I have tried creating iax.conf from scratch
with the same results.
On Mon, 2009-03-09 at 13:32 -0500, Tim Nelson wrote:
Did your iax.conf get overwritten with the upgrade?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Carlos
On Tue, 2009-03-03 at 09:24 -0300, Gustavo A Gonzalez wrote:
Hello, Is there a field into the Zapata.conf file that affect the
value for the fields “start” and “answer” in the CDR? I have this
fields with the same value and I dont understand why. Thanks!!
This is because analog
On Mon, 2009-03-02 at 22:31 +0200, michel freiha wrote:
Hi all,
I'm using asterisk in real time mode...All extensions are defined in
table sip_buddies...Everything looks fine and asterisk is reading
extensions info from the sip_buddies table...The problem occurs as
soon as any information
Where can I find the realtime table definitions for Asterisk 1.6.x?
voip-info.org only has the tables for 1.4 and older. Shouldn't these
tables be in the documentation for 1.6?
I am rebuilding our configuration application for 1.6 and need to know
what changes to make to the
I am using Asterisk 1.6.0.6-rc1 with Mysql CDR. I have a couple extra
fields on the database to store cost information. The problem I have is
that my costing system relies on a certain field being NULL so it knows
that it has not been processed. Since my upgrade to 1.6 I noticed that
Anybody here is able to use Aastra phones with Asterisk 1.6.0.5?
Making calls is not a problem but when you receive a call it always
drops at 1:45 minutes, always!
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
calls.
On Fri, 2009-02-13 at 12:56 -0600, Anthony Messina wrote:
On Friday 13 February 2009 11:39:07 Carlos Chavez wrote:
Anybody here is able to use Aastra phones with Asterisk 1.6.0.5?
Making calls is not a problem but when you receive a call it always
drops at 1:45 minutes
We made a very simple application to insert the cost of a call into the
CDR table that Asterisk uses. We recently upgraded to Asterisk 1.6 and
I noticed that my application stopped working.
The reason is that my application depends on a field called route to
be NULL so that it
I am getting the following warnings on the CLI when loading Asterisk
1.6.1-rc1:
[Feb 12 12:32:34] NOTICE[22261]: timing.c:59
ast_install_timing_functions: Multiple timing modules are loaded. You
should only load one.
[Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders:
I upgraded my office server from 1.4.22 to 1.6.0.5 on the weekend and
after some testing there seems to be a compatibility problem when using
Aastra phones. With 1.6.0.5 all incoming calls to all Aastra phones
were dropped after a minute or so. I installed 1.6.1-rc1 and now the
newer
I upgraded my office PBX from 1.4.22 to 1.6.0.5 so we can start
evaluating and testing. I did not really test it over the weekend, just
made sure I could dial in and out. Today we are finding that incoming
calls to our POTS lines get dropped after a couple of minutes. All I
can see in
This problem only seems to occur when using Aastra phones. Calls to
Polycom never drop. Anyone know of a setting for Aastra that could
cause this?
On Mon, 2009-02-09 at 13:22 -0600, Carlos Chavez wrote:
I upgraded my office PBX from 1.4.22 to 1.6.0.5 so we can start
evaluating
On Fri, 2009-01-30 at 17:17 +, Edwin Quijada wrote:
Hi !
I am trying to connect Asterisk with Avaya Definity.
I use this tutorial to do this
http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html
The comunication between avaya and asterisk is fine but without
Is there a bullet type document with the features each version of
Asterisk has? I know you can read the CHANGES file but that is not
something you give a customer. I just need a one or two page document
with bullet points showing the features added from 1.2 to 1.4 and from
1.4 to 1.6.
Since 1.4.22 realtime status for sip peers seems to be broken. If I do
a sip show peers from the CLI I get this:
2001/2001 192.168.2.234D 5060 UNKNOWN
Cached RT
It is arbitrary which peers will say OK and which will say UNKNOWN and
it changes
I have this call:
SIP/protel-525512047 default 90445528885371 1 Ringing
AppDial (Outgoing Line) 90445528885371 264:24:2
(None)
I cannot use the soft hangup commando from the CLI because I do not
know the whole SIP channel string. What other
Zaptel 1.2.9.1 will not work with Asterisk 1.4.22. I would recommend
you install Zaptel 1.4.12.1 or go to DAHDI. The first thing you need to
do is erase all the zaptel modules from the /lib/modules/kernel
version directory and do a depmod -a to make sure only the new DAHDI
or Zaptel
I recently upgraded a server to Asterisk 1.4.22 with OpenR2.
Previously I was using 1.4.18. It seems that 1.4.22 has a big bug using
chan_alsa.so for overhead paging. After rebooting the server it would
work once or twice and then I just got an error on the CLI:
[Jan 7 10:35:14]
This server is an old installation running on Fedora Core 6. Up to
1.4.18 paging used to work fine. Now the client tells me that they hear a
strange noise and a few seconds of ringing on their PA system. Anyone know a
solution? Is this a bug?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones
Then why use a queue? The purpose of a queue is exactly to keep people
waiting while agents are all busy.
The only way I can see something like what you want is to put a very
low timeout (maybe 10 seconds) so if all your agents are busy then the
caller will get dropped from the
Use the h extension and execute DeadAGI.
On Wed, 2008-12-10 at 18:21 +0100, Martin Tirsel wrote:
Hello,
I am googling for a while but google seems to be broken today, no
solution yet :D I have a PHP script which needs to be started after
Dial() has ended. If there is no answer,
I have a customer running Asterisk 1.4.22 and FreePBX 2.5.0 that is
having problems with Voicemail. They can listen to their voicemail but
on the weekend it stopped delivering messages via email. The only thing
I can notice is that the permissions for the files on teh voicemail
Is there another web front end for meetme apart from Web-MeetMe? Since
it keeps crashing I need a stable solution for a customer. Any
recommendations? Even a commercial app would be acceptable as long as
it is stable and uses Asterisk.
--
Telecomunicaciones Abiertas de México S.A. de
I am having an issue with a Rhino channelbank connected to a Digium
TE411P card. The server has 3 E1 R2 links and the fourth port is used
to connect a Rhino FXO channelbank with 12 lines. The first four ports
on the rhino are GSM adapters. From time to time I can see the channels
301 - 400 of 750 matches
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