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:
2012/7/5, giovanni.v i...@keybits.org:
The matter becomes more difficult approaching a building install as
there are no devices to handle properly that.
I think the snom PA-1 may be a good candidate to play with because of a
versatile I/O that could be interfaced to a custom door-phone bridge
2012/7/5, C F shma...@gmail.com:
snip
no sure if you can have it function
such that any number entered will actually be send to a gateway.
To me, that is the key selling point :
people are used to just dial a number and then press a Fax button that
I can't succeed in anything more than
2012/7/5, C F shma...@gmail.com:
I searched a bit more,
http://www.muratec.com/catalog/F320_config.html#email
The above model supports t.37
That's very interesting to know.
I quickly googled for t.37 and found several other vendors mentioning
this (some from rather old documents).
The strange
I think you are supposed to have a IP based terminal in order to carry both
video and audio.
If budget is accept, it is possible to setup several iOS devices as dedicated
SIP terminals
On Jul 6, 2012, at 8:49 AM, Olivier wrote:
2012/7/5, giovanni.v i...@keybits.org:
The matter becomes
Hello,
Thank you for the clarification.
Just a few questions:
1. What is the Timer1 used for?
2. Since timerb is for all responses, final and provisional, the
comment in sip.conf perhaps should point that out instead of implying
only for provisional responses: If a provisional response is not
Il 06/07/2012 8.49, Olivier ha scritto:
Handling both audio and video seems difficult.
Sure, I don't know a device being able to handle all problems involved
in that project.
Another brick near the wall, but no versatile I/O:
-
Hello,
If you would like to make out bound call (from Asterisk to SIP provider), it is
fine.
But if you want have inbound call (from SIP provider to Asterisk). I think you
are supposed to have something like this
sip.conf
register = 5552530146:your_password@sip3.voipvoip.com/5552530146
Hello,
I have 3 server, 2 running with asterisk and another one generate call
files say some directory callfile/serverA and callfile/serverB (NFS
Sharing) and mounted this directory to respectively on Server A (Asterisk)
and Server B(Asterisk) on /var/spool/asterisk/outgoing.
Server A has
Why don't you just generate call files for each of the servers on the same
server? Anyhow you are not sharing one single pool of call files among servers,
I suspect that's where network drive would come in handy.
Sent from my iPhone
On Jul 6, 2012, at 6:56 PM, Chandrakant Solanki
On Friday 06 July 2012, Chandrakant Solanki wrote:
I have 3 server, 2 running with asterisk and another one generate call
files say some directory callfile/serverA and callfile/serverB (NFS
Sharing) and mounted this directory to respectively on Server A (Asterisk)
and Server B(Asterisk) on
Hello;
Is it possible if I have already asterisk installed on Fedora machine to
install the GUI asterisk now without doing a fresh installation using the
Asterisk Now CD?
Which version of the GUI that should be selected to work with the asterisk
version? For example, if I have asterisk 1.8
There are some appliances that support it is well. But those don't
have a scanner, just thru the computer. MultiTech FaxFinder comes to
mind, for the price they are excellent.
On Fri, Jul 6, 2012 at 3:13 AM, Olivier oza_4...@yahoo.fr wrote:
2012/7/5, C F shma...@gmail.com:
I searched a bit
- Original Message -
Hello;
Is it possible if I have already asterisk installed on Fedora machine
to install the GUI asterisk now without doing a fresh installation
using the Asterisk Now CD?
Which version of the GUI that should be selected to work with the
asterisk version? For
Hi,
I am trying to configure some static queues in asterisk, it's almost
working, the problem is that asterisk is not verifying if the queue has
logged members. For example, if I create queue called test, which has no
members logged in, and try to place a call using Queue(test) I get into
On Friday 06 July 2012, Chandrakant Solanki wrote:
I have set the folder (callfile/Server{A/B}) permission to 777 as well
as call file permission to 777.
On Fri, 6 Jul 2012, A J Stiles wrote:
(By the way, you should have permissions 666 for a callfile, not 777.
Callfiles should not be
On 07/06/2012 10:15 AM, Antonio Modesto wrote:
Hi,
I am trying to configure some static queues in asterisk, it's almost
working, the problem is that asterisk is not verifying if the queue has
logged members. For example, if I create queue called test, which has no
members logged in, and
- Original Message -
It has a Digium Wildcard TE122
If it has an onboard echo canceler, try disabling it and retrying. Just a shot
in the dark, going from my experience with other cards and same symptoms. If
the card is new(ish) I would think Digium could provide support to you for
On Fri, Jul 06, 2012 at 11:10:43AM -0500, Tim Nelson wrote:
- Original Message -
It has a Digium Wildcard TE122
If it has an onboard echo canceler, try disabling it and retrying.
Just a shot in the dark, going from my experience with other cards
and same symptoms. If the card is
On Fri, Jul 06, 2012 at 11:07:26AM -0500, Bill Dunn - VCI Internet Services
wrote:
It has a Digium Wildcard TE122
I've asked Digium about the card below. They say the -1 in the
bipolar and CRC errors is ok. They don't change.
Description AlarmsIRQ
Bill Dunn - VCI Internet Services wrote:
I have an Asterisk server configured to run as voicemail with a T1 and
SMDI.
It has 1.6.1.6 (dahdi 2.1.0.4) and Centos 5.6 and has worked great for a
few
years. I am configuring a new server with Asterisk 1.8.13 (dahdi 2.6.1) on
Centos 5.8
The
On Fri, 2012-07-06 at 11:09 -0500, Kevin P. Fleming wrote:
On 07/06/2012 10:15 AM, Antonio Modesto wrote:
Hi,
I am trying to configure some static queues in asterisk, it's almost
working, the problem is that asterisk is not verifying if the queue has
logged members. For example, if
Thanks Ron. I have had my chan_dahdi.conf file set as follows with the same
result.
[trunkgroups]
[channels]
switchtype=national
usecallerid=yes
callerid=asreceived
cidsignalling=smdi
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
usesmdi=yes
smdiport=/dev/ttyS0
I used the dahdi_monitor to record the audio on the T1 channel of the
working server and the new server. The audio stream of the working server
allowed me to hear the audio I heard over the phone call plus the DTMF at
the very beginning. The audio of the new server was completely messed up.
On 07/06/2012 12:36 PM, Antonio Modesto wrote:
I don't want the users to manually login in the queue, I want they join
the queue when they turn on their phone. I thought that this was the
right way of doing it, how can I do it?
That's a reasonable way to do it if you like, although it's
He's probably using softphones
-Original Message-
From: Kevin P. Fleming kpflem...@digium.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 06 Jul 2012 13:32:20
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Bill Dunn - VCI Internet Services wrote:
Thanks Ron. I have had my chan_dahdi.conf file set as follows with the
same
result.
[trunkgroups]
[channels]
switchtype=national
usecallerid=yes
callerid=asreceived
cidsignalling=smdi
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
On Fri, 2012-07-06 at 13:32 -0500, Kevin P. Fleming wrote:
On 07/06/2012 12:36 PM, Antonio Modesto wrote:
I don't want the users to manually login in the queue, I want they join
the queue when they turn on their phone. I thought that this was the
right way of doing it, how can I do it?
I am planning on building a testing module which would spawn about 500
calls in order to test the performance of the network by transferring
audio/speech files to end points at that juncture.Is it possible to spawn
as many concurrent calls (or nearly concurrent calls) using just call
files.Is
OK, is there asterisk-gui that differs that freepbx? Or Freepbx is a GUI for
asterisk?
In other words, if I have asterisk and I need to add for it a GUI, is there
asterisk-gui which is differs than freepbx or it is the same?
Regards
Bilal
-
Hello;
Is it possible if I have
Hi All;
Based on what I have to use Trixbox or FreePBX?
Can someone advise?
Regards
Bilal
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New to Asterisk? Join us for a live introductory webinar
Hi there.
i am seriously stuck with an asterisk and sip problem.
the following sip.conf works with respect to some_peer:
[general]
bindaddr = x.y.z.w
nat = no
[some_peer]
type=peer
host=somehost
secret=somesecret
some other
unrelated options
here x.y.z.w is the ip address of the interface
On Fri, Jul 06, 2012 at 01:28:01PM -0500, Bill Dunn - VCI Internet Services
wrote:
I used the dahdi_monitor to record the audio on the T1 channel of the
working server and the new server. The audio stream of the working server
allowed me to hear the audio I heard over the phone call plus the
- Original Message -
OK, is there asterisk-gui that differs that freepbx? Or Freepbx is a
GUI for asterisk?
In other words, if I have asterisk and I need to add for it a GUI, is
there asterisk-gui which is differs than freepbx or it is the same?
There have been a handful of other
- Original Message -
Hi All;
Based on what I have to use Trixbox or FreePBX?
Can someone advise?
Trixbox includes FreePBX as it's GUI. However, keep in mind it is a
bastardized, forked version of FreePBX that has seen nary any new development
or innovation in some time. At this
I did one try and did not hangup:
I called the extension and when the voicemail answered, I did not leave any
message and stayed waiting .. waiting .. waiting ..
It did not hangup from it self ! How much it stay waiting me to leave a message?
Why I am trying this? OK, the answer is: Because I
I haven't used it, so can't recommend it per se; but as I understand
it, iperf is a tool that can do that kind of simulation for you:
http://iperf.sourceforge.net/ might be worth trying before you build
your own modules.
Regards,
Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729
On
I've previously used iperf for my project and It can only simulate TCP/UDP
traffic.and the thing is I'm testing this on a platform which does only
RTP/SIP.I am not sure they've this facility.Anyhoo i wanted to know if it
was possible to make such concurrent calls using Asterisk
On Fri, Jul 6,
Dears;
Thanks for all the replies and help.
First of all, I am not looking to have the custom context only for outbound, I
need this also to separate the extensions into partitions, so I can have same
extensions in different contexts, also extensions in context A can not call
extensions in
- Original Message -
From: bilal ghayyad bilmar...@yahoo.com
To: asterisk-users@lists.digium.com
Sent: Friday, July 6, 2012 4:56:02 PM
Subject: Re: [asterisk-users] FreePBX: How to hangup if the caller did not
press # after the voicemail message
I did one try and did not
Hi Tim,
How about AsteriskNow?
Thanks and BR,
Anam
On 7/7/12, Tim Nelson tnel...@rockbochs.com wrote:
- Original Message -
Hi All;
Based on what I have to use Trixbox or FreePBX?
Can someone advise?
Trixbox includes FreePBX as it's GUI. However, keep in mind it is a
bastardized,
- Original Message -
Hi Tim,
How about AsteriskNow?
Thanks and BR,
Anam
On 7/7/12, Tim Nelson tnel...@rockbochs.com wrote:
- Original Message -
Hi All;
Based on what I have to use Trixbox or FreePBX?
Can someone advise?
Trixbox includes FreePBX as it's GUI.
Opss, sorry not read it carefully :((
On 7/7/12, Tim Nelson tnel...@rockbochs.com wrote:
- Original Message -
Hi Tim,
How about AsteriskNow?
Thanks and BR,
Anam
On 7/7/12, Tim Nelson tnel...@rockbochs.com wrote:
- Original Message -
Hi All;
Based on what I have to
Hi,
I have 100+ call file generated in other directory, and by using program, I
have moved 10-10 files in /var/spool/asterisk/outgoing, and call made
successfully.
Once all call completed, I found following error for all files...
[Jul 7 10:54:03] WARNING[7884]: pbx_spool.c:407 scan_service:
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