Hello,
We've purchased a 2nd Wildcard TDM2400P for our asterisk server. I've looked
around but I can't find documentation on how to properly setup 2 or more
telephony boards in asterisk. In particular, I want to ensure that the boards
are assigned the same device channels on boot (for example,
Hi,
I have an odd situation where one certain user presses 7 to delete a
voicemail. She gets a response saying that the message has been deleted, but
it actually gets moved to her Old box. I have deleted her entries out of the
config files and added them back. This seemed to correct the
Is it possible to load phone registration information stored in sipfriends
MySQL DB, so that Asterisk thinks those phones are already registered?
This would be very usefull for a redundant server...
Regards,
Ricardo Carvalho.
___
--Bandwidth and
I'm in the process of writing a simple autodialer to dial a list of numbers
and play a message. One of the options I want to give them is a way to
dial X to have a customer service representative call you
Looking for a simple way to pass the number that I dialed to a script in
I am attempting to use a Mediatrix 1204 to interface to multizone paging
from Asterisk. I have 4 different paging interfaces and want to connect
each of those 4 to an FXO port on the Mediatrix. The desired result is
to be able to issue some SIP dial string from asterisk, seize the FXO
port on
Part of a supervisor menu I'm writing requires that I allow the
supervisor to choose to ChanSpy a channel from the main menu then return
back to the menu to choose other options when she's done. Is there a
way to 'exit' ChanSpy and continue down the dialplan? Or is a caller
stuck in ChanSpy
Nice!
On 7/11/07, Guillermo Salas M. [EMAIL PROTECTED] wrote:
On Wed, 2007-07-11 at 08:32 -0400, Maximo Villamayor wrote:
you can prove this www.portsip.com
You can use the older version of firefly that supports IAX2/SIP
protocols and g729 codec.
Get the sofhophone and codec from:
Asterisk gurus,
To have analog call progress (which as far as I know asterisk does not
have right now)
does it not come down to 4 states to detect - and I hope that some
asterisk gurus can
implement those 4 states easily.
I see this as 4 states:
Busy - a 50% duty cycle
Ringing - a % on duty
Hello all,
A client of us, needs a queue monitoring system. In realtime he needs to now
the PRI status, the agents logged in and logged out, the number of received
calls by agent, ,etc.
I am not a call center specialist and i want to find a call center software
to offer to my client that
I can use a USB modem with asterisk to connect to the
PSTN network right? It'll serve the same functionality
as an FXO card? Also, any idea if I can get these
modems with mutiple ports (12 or 24)?
Thanks,
Doug
MCelo wrote:
Why don´t you try to change this line:
qualify=yes
to this one:
qualify=3000 or 4000
Also, please make sure you are using version 1.4.7.1 of Asterisk or
later if you are using the 1.4 series. I have fixed some bugs in the
1.4 branch related to this since 1.4.6.
--
So I'm back on this matter (I thus give enough time for a good samartan to
help), and now, I think that PlayDTMF is not designed for what I want to do
:
I wanted to simulate pressed keys in order to give order to Asterisk (an
attented xfert) ; e.g. generate a #2 (it is my combination to
On Wed, 2007-07-11 at 21:57 -0600, Al lists wrote:
Nice!
This version supports IAX2 and SIP. Windows users will be happy using
it ;)
Regards,
You can use the older version of firefly that supports
IAX2/SIP
protocols and g729 codec.
Are you doing *0 after you flash the hook? This will flash the fxo
line for you. I do wish there was a way to get Asterisk to answer the
call waiting on the fxo, all I ever get is the call waiting beep and I
get to answer it myself, otherwise it goes to telco's voicemail.
on Wednesday
You might want to have a look at QueueMetrics: http://queuemetrics.loway.it/
I am not sure if it supports all features you are looking for but it
should be a good start.
=Stefan
--
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221
Dnia 2007-07-11, o godz. 09:42:03
Stephen Bosch [EMAIL PROTECTED] napisał(a):
Hi, Jakub:
Hello
Is this normal behaviour? How to change this?
Sadly, this is normal behaviour.
Log says everything, MOH should stop after call pickup, not before
Dial.
Well, no -- here, the log shows
On 7/11/07, Jakub Głazik [EMAIL PROTECTED] wrote:
Asterisk [EMAIL PROTECTED]
Client hears pure silence when waiting for call answer. Music on hold
stops
when transferer pics a number and client doesn't even hear ringing.
Is this normal behaviour? How to change this?
Log says everything, MOH
Barry,
I'm not sure I understand but I think you actually want a Mediatrix
1104, which has 4 analog device (phone, fax, etc) ports, to provide dial
tone to your paging system. I'm assuming your 4 paging systems are the
equivilent of 4 phones..ie, subscriber devices. They would each get an
On 7/11/2007 at 11:04 AM, Joe acquisto [EMAIL PROTECTED] wrote:
Since the beginning (of my Asterisk life) I have an install that is,
supposedly, set up for call waiting.
Using a TDM400p, with FXO and FXS modules.
On the Analog phones, I can hear the Incoming call (call waiting) tone, but
On Wed, 11 Jul 2007, Jadrien Wauthier wrote:
Hi,
I have an odd situation where one certain user presses 7 to delete a
voicemail. She gets a response saying that the message has been
deleted, but it actually gets moved to her Old box. I have deleted
her entries out of the config files
On 7/3/07, Exploding Lemur [EMAIL PROTECTED] wrote:
I'm using Asterisk 1.4.5 (will try 1.4.6 on Thursday, but I don't see
anything in the changelog after the 1.4.5 release dealing with
distinctive ring), zaptel 1.4.3, and wanpipe 2.3.4-10 with a Sangoma
A200 card. I enabled
Hello !!!
I'm trying to setting up my DVG-2032S Voip Gateway to use Asterisk as
outbound proxy, that's because I already have this gateway before to begin
to play with Asterisk.
Every time when I enable the OutBound Proxy option and call from my
Ericsson PBX I got the follow message in
Hi -
I have a strange comportment of the MOH system on my asterisk.
When i respond to a call and after fews second i set this call in hold
mode the correspondent listen the music fine.
When i re-take my correspondent at T0 instant the music is paused. And
when i re-hold him at T60 (60 second
Is the list up? I haven't gotten mail in the last 24 hours.
Alex
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi!
I want to have this behabior in my queue.
When a call come in, if there are unavailable agents or and busy agents, the
queue reject the call.
Thanks !
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing
try to transfer current call to parking spot, i.e. exten 700, then deal
with new incoming call, then go back to parking space to pick up old
caller when you're free. Just set the parking extension timeout to
something long so they don't fall out right away.
Moj
Joe acquisto wrote:
On
Asterisk gurus,
To have analog call progress (which as far as I know asterisk does not
have right now)
does it not come down to 4 states to detect - and I hope that some
asterisk gurus can
implement those 4 states easily.
I see this as 4 states:
Busy - a 50% duty cycle
Ringing - a % on duty
Doug Zingel wrote:
I can use a USB modem with asterisk to connect to the
PSTN network right? It'll serve the same functionality
as an FXO card? Also, any idea if I can get these
modems with mutiple ports (12 or 24)?
WRONG!
___
--Bandwidth and
No. A USB modem will *not* work as a FXO card. You're thinking of the
X100P card which was a rebranded Intel modem. These have been
discontinued by Digium and only third party suppliers still sell them.
But taking a normal modem and using it as a FXO will not work (most
modems do not pass
Hello !!!
I'm trying to setting up my DVG-2032S Voip Gateway to use Asterisk as
outbound proxy, that's because I already have this gateway before to begin
to play with Asterisk.
Every time when I enable the OutBound Proxy option and call from my
Ericsson PBX I got the follow message in
I decided to try 1.4 today. Zaptel builds, installs and seems to load
correctly and Asterisk builds, installs and works but no Zap lines.
When I typed make menuselect for Asterisk and went poking around I
discovered that chan_zap is marked XXX and I can't figure out why
that might be. Any
equis software wrote:
Hi!
I want to have this behabior in my queue.
When a call come in, if there are unavailable agents or and busy agents,
the queue reject the call.
Thanks !
I think you need checkout:
Introduced right after the v1.0 release
If you wish to remove callers from the
equis software wrote:
Hi!
I want to have this behabior in my queue.
When a call come in, if there are unavailable agents or and busy agents,
the queue reject the call.
Thanks !
Also check out
joinempty=strict
...it's in the same article:
On 7/12/07, Stefan Reuter [EMAIL PROTECTED] wrote:
You might want to have a look at QueueMetrics:
http://queuemetrics.loway.it/
I am not sure if it supports all features you are looking for but it
should be a good start.
QueueMetrics is working well for me in a 75 seat call center, but it
I am having a problem with my asterisk gateway, it is accepting only G729,
the client is offering G729 and G723.1, however for some reasons, around 15%
of calls are rejected due to failed codec negotiation giving an codec error
No compatible codecs, not accepting this offer.
Anyone gone through
Is it possible to have Asterisk allow the From address in a SIP invite
to be different to the required digest username?
The auth parameter supposedly allows it, but whether or not I set auth
to be what the UA sends as the digest username, Asterisk just complains
that the from and the digest are
We haven't taken a poll, so I can't say that most users are having no
trouble, but I do know that several users have expressed concern about
delayed messages.
Although Dimitri's plea to the 'Admin' may not have been carefully worded,
it would be nice to have some feedback on this issue. As far as
I try setting joinempty=strict, but if there are all agent busy answering a
call, i need that the queue reject new callers until one of the agents ends
their calls.
Example:
Agent1 on call
Agent2 on call
When one call arrive, the queue reject the call.
I don´t want to have callers waiting in
On 7/12/07, Ricardo Carvalho [EMAIL PROTECTED] wrote:
Is it possible to load phone registration information stored in sipfriends
MySQL DB, so that Asterisk thinks those phones are already registered?
This would be very usefull for a redundant server...
Look at realtime sip
should help you
On Thu, 2007-07-12 at 13:26 -0400, Lee Jenkins wrote:
equis software wrote:
Hi!
I want to have this behabior in my queue.
When a call come in, if there are unavailable agents or and busy agents,
the queue reject the call.
Thanks !
Also check out
joinempty=strict
So who do you pay to use the G723 codec?
Best regards,
Al Bochter
http://www.BochterServices.com
---
See what we are selling at auction
http://www.epier.com/auctions.asp?bochterservices
On 7/12/07, O. Kamal [EMAIL PROTECTED] wrote:
I am having a problem with my asterisk gateway, it is accepting only G729,
the client is offering G729 and G723.1, however for some reasons, around
15% of calls are rejected due to failed codec negotiation giving an codec
error No compatible codecs,
On Thu, 2007-07-12 at 12:54 +, lemmel lemmel wrote:
So I'm back on this matter (I thus give enough time for a good samartan to
help), and now, I think that PlayDTMF is not designed for what I want to do
:
I've done this, and yes, you are correct, PlayDTMF is the wrong thing,
it goes in
Ira wrote:
I decided to try 1.4 today. Zaptel builds, installs and seems to load
correctly and Asterisk builds, installs and works but no Zap lines.
When I typed make menuselect for Asterisk and went poking around I
discovered that chan_zap is marked XXX and I can't figure out why
that
Don Kelly wrote:
Although Dimitri's plea to the 'Admin' may not have been carefully worded,
it would be nice to have some feedback on this issue. As far as I can tell,
there has been no response by moderators/admin to the messages regarding
list delays, including messages that I have sent
On Thu, 2007-07-12 at 15:03 -0300, equis software wrote:
I try setting joinempty=strict, but if there are all agent busy
answering a call, i need that the queue reject new callers until one
of the agents ends their calls.
Example:
Agent1 on call
Agent2 on call
When one call arrive, the
On Thu, Jul 12, 2007 at 10:01:41AM -0700, Ira wrote:
I decided to try 1.4 today. Zaptel builds, installs and seems to load
correctly and Asterisk builds, installs and works but no Zap lines.
When I typed make menuselect for Asterisk and went poking around I
discovered that chan_zap is
Hi
I have asked this questions,but have no answer :) I also want Asterisk do
not check to head with digest username in registration,how can we do that?
On 7/12/07, Douglas Garstang [EMAIL PROTECTED] wrote:
Is it possible to have Asterisk allow the From address in a SIP invite to
be different
On Thu, 2007-07-12 at 14:39 -0400, Al Bochter wrote:
So who do you pay to use the G723 codec?
It's possible to use the G.723.1 codec with Asterisk by buying a Digium
TC400B transcoder card[1]. Without that card, the best Asterisk can do
is to pass through the packets, but it can't doing any
On Thu, 2007-07-12 at 12:51 -0600, Steve Murphy wrote:
On Thu, 2007-07-12 at 12:54 +, lemmel lemmel wrote:
So I'm back on this matter (I thus give enough time for a good samartan to
help), and now, I think that PlayDTMF is not designed for what I want to do
:
I've done this, and
Alex Roston wrote:
Is the list up? I haven't gotten mail in the last 24 hours.
Alex
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Bloody good question.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Ma
Sent: Thursday, July 12, 2007 1:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Different SIP From and Auth?
Hi
I
Hello,
We're running into a problem I thought some of the enlightened people on
this list might be able to help with. Our VoIP stuff has grown to the
point where it makes sense to get a PRI (we've been doing things purely
voip till now).
The problem we're running into is this:
We have
Hi,
I want to make sure I'm clear on something.All of the phone
systems we have setup have been either analog lines, or PRI lines. We
have a customer who is going to use a T1 line with 4-digit outpulse.
Do I need to do anything special to get Asterisk to recognize the
outpulse, or will it
Carlos Chavez wrote:
On Thu, 2007-07-12 at 15:03 -0300, equis software wrote:
I try setting joinempty=strict, but if there are all agent busy
answering a call, i need that the queue reject new callers until one
of the agents ends their calls.
Example:
Agent1 on call
Agent2 on call
Same issue in 1.2.21 on FC6 with mpg123 0.66 (0.59r would not install on FC6).
What version of mpg123 are you using? What about Linux?
Here is my musiconhold.conf
[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3
there are mp3 files in the specific directory, and they are the default
On Wed, Jul 11, 2007 at 09:34:51AM -0600, Stephen Bosch said:
Walt Reed wrote:
No, as I explained before with the reasons why, please don't post them
here. Send them DIRECTLY to the list admins. It is 100% off topic to
keep discussing a list administration / mail delivery problem here.
Hello Everyone,
I have just released AstLinux 0.4.6.1. The only difference between
the last binary release (0.4.5) and 0.4.6.1 is Asterisk 1.2.21.1. As
always, AstLinux can be downloaded at the SourceForge project page:
http://sourceforge.net/projects/astlinux/
Thanks!
--
Kristian
On 7/6/07, James FitzGibbon [EMAIL PROTECTED] wrote:
Under 1.4.5 and 1.4.6, I've seen a few instances where the channel name in
queue log entries is replaced by a snippet of a manager event:
Nobody else seeing this? I'm at a loss - it's only one queue now that I go
and look at the history,
On 7/11/07, Guillermo Salas M. [EMAIL PROTECTED] wrote:
On Wed, 2007-07-11 at 08:32 -0400, Maximo Villamayor wrote:
you can prove this www.portsip.com
You can use the older version of firefly that supports IAX2/SIP
protocols and g729 codec.
Get the sofhophone and codec from:
Tomislav Parcina wrote:
There is hotel application weary popular in Croatia - Micros-Fidelio.
Now I need to connect Asterisk with this application for purpose of
billing. Thing is that hotel would like to give customer one bill for
every service that he used while he was in hotel.
Has
Kevin P. Fleming on 12 July 2007 19:52 wrote:
8 SNIP! 8-
we haven't yet found the root cause of
the delays, but it does appear to be a large number of subscriber
addresses that fail to resolve via DNS (but 'soft failures' (timeouts),
not hard failures).
As we get through the
Hello Voipcrazy,
It's funny you should mention that - we've just released (as in today) a FREE
version of our OrderlyStats service for call centre and queue monitoring and
management.
OrderlyStats features realtime (synchronous/message-based) display of all
queue, agent and caller events so
Walt Reed wrote:
On Wed, Jul 11, 2007 at 09:34:51AM -0600, Stephen Bosch said:
Walt Reed wrote:
No, as I explained before with the reasons why, please don't post them
here. Send them DIRECTLY to the list admins. It is 100% off topic to
keep discussing a list administration / mail delivery
Hi all, i am having a major asterisk problem. I think it started around
1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we
start getting busy signals, all our 4 line hunt group is busy, i then
check the channels and there are open calls that were hung up long ago.
i thought
Newbie question(s):
From what I can determine it sounds like the SMS messaging isn't as
robust as it could be (?). I'm wondering if there's active work on
that right now or if it's more of an issue about PSTN carrier that
one would be using who would be responsible for passing the
On Thu, 2007-07-12 at 17:54 -0400, James FitzGibbon wrote:
On 7/6/07, James FitzGibbon [EMAIL PROTECTED] wrote:
Under 1.4.5 and 1.4.6, I've seen a few instances where the
channel name in queue log entries is replaced by a snippet of
a manager event:
Should I just
Most of the users using this list do not experience the issue
you are having, rather than insult the admins, please trouble
shoot and if you cannot, at least post headers so others can.
Received: from lists.digium.com ([216.207.245.17]) by
nlpiport01.prodigy.net.mx
with ESMTP; Thu, 12 Jul
On Mon, 2007-07-02 at 21:56 +0530, ram wrote:
still i need to re-register or
just copy the g729 of 1.4 and copy the license
will this work?
The only time you should need to re-register the g729 license is when
you change network cards ... any network card on the box running
asterisk, this
69 matches
Mail list logo
