There was a bit of traffic on this list a while ago regarding OPTUS
multi line that comes in via a DSL box, and I am hoping some of those
people are still hanging around, and solved their problems.
We apparently have a 14B channel service with optus.
I have been trying to configure Asterisk,
There was a bit of traffic on this list a while ago regarding OPTUS
multi line that comes in via a DSL box, and I am hoping some of those
people are still hanging around, and solved their problems.
We apparently have a 14B channel service with optus.
I have been trying to configure Asterisk,
Hello,
routing based on DNIS is dependant on what your telco sends you. Usually on
Robbed-bit T1s(RBS) they will send you ANI and DNIS together separated by
stars like this:
*7275551212*1234*
(where 7275551212 is the ANI[callerID] and 1234 is the DNIS[last 4 digits of
the number dialed])
In
What kind of call center: inbound, outbound or both?
how many lines per agent will you have?
what kind of trunks will you be using?
do you need to tie into an existing database?
do you want screen-pops?
MATT---
-Original Message-
From: Zeeshan [mailto:[EMAIL PROTECTED]
Sent: Tuesday,
---
-Original Message-
From: Zeeshan [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 02, 2005 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call center 20 seats
mattf wrote:
What kind of call center: inbound, outbound or both
astGUIclient is not a configuration tool, it is an end-user-interface that
extends the functionality of your phone through a web browser.
We recommend AMP if you need a web-based config utility.
MATT---
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Friday,
I don't notice it on my TDMoE that is configured as PRI either. Looks like
you need to post a bug to the tracker.
MATT---
-Original Message-
From: Weezey [mailto:[EMAIL PROTECTED]
Sent: Monday, July 11, 2005 4:33 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TDMoE and
Hello,
We've just started using TDMoE(local T1s connecting between Asterisk servers
in the same building over the LAN) to connect a few of our high-availability
servers instead of using crossover T1 cables. The 3 servers we have
connected to each other over TDMoE are running just fine and we have
Hello,
I would recommend Slackware mostly for it's streamlined, minimalist approach
and history of stable distro releases. But with that said, the most
important thing is building a custom streamlined Linux kernel no matter what
distro you use. This can save you bootup time as well as speeding up
You can send both paties to a meetme conference with Manager Redirect. Or if
you are feeling more adventurous you could load the Manager Bridge patch
that I posted to the bugtracker two months ago. It allows bridging of any
two existing channels together through a manager action:
Hello,
Either Digium or Sangoma can overnight a card to you. As the car drives you
could go to Toronto and pickup a card from Sangoma if you needed if a few
hours before Overnight would deliver it.
There are also a lot of resellers that can overnight to you as well.
MATT---
-Original
Hello,
Need to give some more info here:
- What kind of hardware?
- What distro/version of Linux?
- What version of Asterisk?
- What wanpipe driver version for Sangoma card?
- What firmware version for Sangoma a104 card?
- What are your zaptel/zapata settings for this machine and your other
Hello,
We've released another update to our Asterisk GUI Client suite: 1.1.4
http://astguiclient.sf.net/
The client suite runs on Windows, UNIX and Mac, includes the VICIDIAL
auto-dialer and is free as in GPL.
(the suite is not an asterisk configuration tool)
This package is geared towards
We have two Baytech RPC3 remote power switches(8 outlets each), they are
great, you can telnet into them and reset ports as needed. I even setup one
of them to be controlled by an AGI script on our Asterisk servers to cycle
power over the phone. Saved countless hours of driving. APC makes them too
Let me throw another complaint against XO on the table. They actually shut
off the wrong T1 and they transferred all of the DIDs to the T1 they shut
off! how screwed up is that? We are now about 2 years later and their
billing department still calls us every month for nonpayment of the T1 that
] On Behalf Of mattf
Sent: Saturday, June 11, 2005 9:04 PM
To: 'Brian Roy'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Opinions of Sphinx?
We use batch sphinx to analyze recordings at night. We attempted
real-time
sphinx, but it is way too slow and resource
In our experience, the total cost of softphones(money, reduced sound quality
and lower reliability) in a large call center environment is actually
greater over time than the cost of a channelbank and cheap analog
headphones. We've tried 2 softphones, 2 kinds of SIP VOIP hardphones, 2
kinds of SIP
We use batch sphinx to analyze recordings at night. We attempted real-time
sphinx, but it is way too slow and resource-intensive to use for realtime on
Asterisk with more than a couple lines at once(and that's at the poor
quality settings). We have not tried sphinx4, but I wouldn't imagine that it
Hello,
This issue was just handled Monday on the astguiclient-users list:
http://sourceforge.net/mailarchive/forum.php?thread_id=7448401forum_id=4358
6
You just need to use OLD_PASSWORD in the SET PASSWORD for your mysql server
to get the auth method for that account back to the pre 4.1.12
Hello,
Several people asked to get a hold of the stats I used to determine that the
Sangoma T1/E1 boards performed better in our real-world tests than the
Digium boards. We've decided to post our results after confirming them over
the past month of operations.
Here is a link to the last two
Hello,
I'm the lead developer of astGUIclient(with VICIDIAL) and I tried GnuDialer
a little while ago. It is different in several ways from how VICIDIAL
operates:
- Gnudialer is partially compiled into Asterisk and uses Asterisk agents
while VICIDIAL operates entirely on top of a stock Asterisk
Hello,
We've released another update to our Asterisk GUI Client suite: 1.1.1
http://astguiclient.sf.net/
Screen shots: http://astguiclient.sourceforge.net/screenshots.html
The client suite runs on both Windows and UNIX, includes the VICIDIAL
auto-dialer and is free as in GPL.
(the suite is not
Need to provide a little more info:
What's the bus speed?
What kind of motherboard would you use with each?
What kind of RAM at what speed?
What cache size are on the CPUs?
Also, what price are these as equals? I've seen two Xeon 2.8GHz 800MHz
processors for about US$450 and a single P4 at the
Hello,
We use astGUIclient, it does have server side apps that have to be installed
on your Asterisk server, but it does have callerID popups that allow you to
search a customizable web page when a call comes in. We are also releasing a
new version of the astGUIclient app next week that is
The method we use for web popups on incoming calls in the astGUIclient
client app that we are working on for release next week is to use
AJAX(Javascript + XMLHTTPRequest) It works in Firefox and IE5+ and doesn't
require any META refreshes. We've been using this internally for the last
month and it
Are you talking about an Asterisk configuration GUI that would modify
Asterisk settings or an end-user GUI that would compliment a regular user's
phone?
MATT---
-Original Message-
From: Mitchel Constantin [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 25, 2005 1:59 PM
To: Asterisk Users
to be extremely helpful when it comes to training.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Monday, May 23, 2005 6:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Inbound call center - reliability
List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Inbound call center - reliability \
scalabil ity with queues
Matt,
Are you doing any call recording / monitoring? What percentage?
Ilan
On 5/23/05, mattf [EMAIL PROTECTED] wrote:
For an inbound call center with 4 T1s and 30-50
with the T1's, when they
receive the call it will be SIP VOIP. There will be media gateways (i.e.
cisco media gateways) to change all T1 signals to VOIP before it reaches the
PBX.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Tuesday, May 24
For an inbound call center with 4 T1s and 30-50 agents on you would do just
fine with a single, one-processor machine. We have handled more than this on
a single P4 server although we use astGUIclient instead of Asterisk queues,
but the load is very similar. I would recommend a Sangoma Quad T1
This may be somewhat of a cross-post with -dev, but I have Manager Bridge
Action working under CVS_HEAD and releases 1.0.6-7.
http://bugs.digium.com/view.php?id=4297
MATT---
-Original Message-
From: Josiah Bryan [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 27, 2005 1:53 PM
To:
What exactly are you looking for?
There are basically 3 commercial solutions: Aheeva, DACX and Sinedialer
and there are 2 open-source solutions: ShadyDial and VICIDIAL
What features do you need that are not addressed by one of these?
MATT---
-Original Message-
From: Anton Krall
Dialers
I like vicidial's features but Im looking for a friendlier user interface..
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of mattf
|Sent: Miércoles, 11 de Mayo de 2005 01:22 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion
Of mattf
|Sent: Miércoles, 11 de Mayo de 2005 03:53 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Predictive Dialers
|
|Could you let us know what you would consider a 'friendlier
|user interface'?
|
|MATT---
|
|-Original Message-
|From: Anton
Hello,
Have you tried the wanpipe-beta8c-2.3.3.tgz release in the custom/2.3.3 dir
on their FTP site? Also, have you contacted Sangoma for support? They are
very responsive.
I am using wanpipe-beta8a-2.3.3.tgz and it's been working great on my A104
for a week now.
MATT---
-Original
Hello,
Have you contacted Sangoma about this? What version of wanpipe and what
version of zaptel/asterisk are you using?
MATT---
-Original Message-
From: Nguyen Trung Tin [mailto:[EMAIL PROTECTED]
Sent: Monday, May 09, 2005 1:25 AM
To: asterisk-users@lists.digium.com
Subject:
My Sangoma Experience in Asterisk: Followup
2005-05-06
original review can be found at:
http://astguiclient.sourceforge.net/Sangoma_experience.txt
It's now been a month since I finished my last round of tests with the
Sangoma A104u board in Asterisk. I have had a lot of conversations with
They would also be competing indirectly with themselves(Cisco and Avaya
license their technology in their phones). But the real reason comes down to
support. I've been following the Polycom/Asterisk thing for 2 years now and
it really comes down to the fact that they do not want Asterisk users
: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 05, 2005 6:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom Images
mattf [EMAIL PROTECTED] writes:
They would also be competing indirectly with themselves(Cisco and Avaya
license
Look at the app_conference description on the wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20app_conference
I believe it does what you want to do, but I really don't know if it works
with CVS_HEAD or stable releases. I'd be curious to hear how it affects
performance as well.
Hello,
is this how you are starting up:
1. modprobe zaptel
2. wanrouter start
3. ztcfg -v
4. asterisk -vvvgc
Also, what version of the wanroute driver software are you using?
MATT---
-Original Message-
From: Chris Mason (Lists) [mailto:[EMAIL PROTECTED]
Sent: Saturday, April 30, 2005
and the distribute the load
and/or intelligence on other Asterisk boxes to connect SIP agents and
all dialing rules, etc?
Thanks,
Daniel
On Apr 28, 2005, at 9:17 PM, mattf wrote:
You can throw together a single P4 3GHz with 1GB RAM and 2 x 80GB SATA
HD
for about $600. One of those can easily handle
I would suggest opening up a bug on the tracker, if it hasn't been done
already, for all of these discussions to be logged to.
From my glancing at the code, I think there would be a little more cleanup
involved in it than just the ast_bridge_chan function. But either way being
part of the Manager
I have never been able to do more than 50 concurrent recordings with Zap -
SIP phone calls without the audio skipping and/or breaking up. Also, if you
are using Digium TE4XXP and want to do a lot of recording I would recommend
against a SCSI RAID card because of the interrupt conflicts that you
server separately several times
a day :) - don't record to NFS mounted drive.
Thanks,
Daniel
On Apr 28, 2005, at 6:42 PM, mattf wrote:
I have never been able to do more than 50 concurrent recordings with
Zap -
SIP phone calls without the audio skipping and/or breaking up. Also,
if you
Maybe this would best be explained in a diagram:
1). person A --- music on hold and person B --- music on hold
2). *some manager API action*
3). person A --- person B
This is what I think he's asking about, how do you take two parties on
different conversations and put them
, April 27, 2005 12:44 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Redirect two channels to each other?
In article
[EMAIL PROTECTED],
mattf [EMAIL PROTECTED] wrote:
Maybe this would best be explained in a diagram:
1). person A --- music on hold and person B
Tunisia should be +1 from GMT
http://www.freedomphones.net/phone_codes_GMT.txt
MATT---
-Original Message-
From: Samuel T. Cossette [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 27, 2005 1:07 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] tonezone in tunisia
Hi,
Doesn't work that way, you have to know the exact channel that you want to
Monitor on SIP. What you can do is a Command Show Channels in the manager
and parse through the output to find the first channel then do your Monitor
command.
MATT---
-Original Message-
From: Dana Olson
failure with 2 Digium TE405P
car ds
Seems odd, though I would suspect the boards.
Have you tried higher end boards, like compaq proliant servers?
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Monday, April 18, 2005 11:25 AM
Title: Vici Dialer
It's
quite easy to do multiple campaigns on VICIDIAL(just create a new campaign, load
leads, assign the leads to that campaign and have the agent log into the new
campaign), you might want to post you question on the astGUIclient-users mailing
list though:
Hello,
I have spend a long time trying to figure out exactly what is the problem
with one of my Asterisk servers, it is the only one at any of our locations
that has two Digium quad T1 cards in it with 7 T1s connected to it. Most of
the rest of our Asterisk servers run identical hardware except
) with the
crossovers.. Unless they are all on the same power grid and protected
I would blame them. my two cents...
On 4/18/05, mattf [EMAIL PROTECTED] wrote:
Hello,
I have spend a long time trying to figure out exactly what is the problem
with one of my Asterisk servers, it is the only one
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Monday, April 18, 2005 10:35 AM
To: 'asterisk-users@lists.digium.com'
Subject: [Asterisk-Users] Motherboard failure with 2 Digium TE405P cards
Hello,
I have spend a long time trying to figure out exactly what is the
problem
with one of my
Keep on bugging the Sangoma guys, I know they are working on several RBS T1
issues right now(They called me Friday to go over a few things) They just
need help from users like you and I to find the bugs in their drivers.
Have you tried any other signalling types other than LOOP?
MATT---
off from 127.0.0.1
So it appears that my variable ${confNo} is not being set, or at least
honored.
Any thoughts?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Thursday, April 07, 2005 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial
My Sangoma Experience in Asterisk: 2005-04-07
Having pushed my Digium Asterisk systems to their capacity many times and
figuring out the limits of the Digium hardware I decided it was time to test
an Asterisk-compatible Sangoma Quad T1/E1 card(AFT-A104u) to see if they
live
As an Asterisk-related Open-Source project developer I would very much like
this idea :)
We could have a competition that ends yearly during Astricon at which point
the application is chosen.
But who would judge which is best?
I'm pretty sure that the front runners if this was done this year
Several of these RBS T1s have been here for many years and before we moved
to Asterisk a few pieces of phone hardware we used were not PRI-compatible.
There is also the fact that we still use Channel banks which are also RBS.
We have started a long process of switching to PRIs as our RBS T1
Hello,
This would be software since I still don't see the Digium echo-cancellers
anywhere for sale and don't know how to get one. If Digium wants to send me
one I would gladly test it.
The overall machine load is also affected by the way interrupts are used and
the fact that Sangoma uses
just create an extension that plays the message and hangs up and use the
manager interface to drop it into the meetme room.
Let me know if you would like an example and I'll whip one up.
We do this kind of thing in astGUIclient to play DTMF tones automatically in
meetme rooms.
MATT---
A sample would be great. I'm hoping that the Official MeetMe2
will have provisions for this, but until then I'll have a
fully functional scheduler.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Thursday, April 07, 2005 3:31 PM
We target Sipura because they are relatively a small company, the core
developers at Sipura used to work for Cisco and worked on their ATA product
before they started their own company. A small company is much more likely
to try something new with little lead-time. Also access to decision-makers
Hello,
I need to correct myself on one of the points I made in my reply last night.
As a very polite developer from Sangoma stated to me(with evidence I might
add)they have in the past and continue to today contribute code to GPL
Asterisk. It doesn't say so on their website but their developers
Here's an idea, Digium buys Sangoma with the massive amounts of cash they
are getting from venture capitalists and just integrate Sangoma designs into
their boards. Not sure how Sangoma would feel about this idea though.
MATT---
-Original Message-
From: Matthew Boehm [mailto:[EMAIL
.
On Thu, 31 Mar 2005 11:00:48 -0500, mattf [EMAIL PROTECTED] wrote:
Here's an idea, Digium buys Sangoma with the massive amounts of cash they
are getting from venture capitalists and just integrate Sangoma designs
into
their boards. Not sure how Sangoma would feel about this idea though.
MATT
You're going to have to go a little more in depth into what you are doing in
this call center.
- Are you going to be doing inbound or outbound? (if so how much of each)
- What kind of phones are you planning on using?
- What is the maximum number of concurrent conversations you plan on having?
-
I've been trying to get a test G5 in our office from Terrasoft for the last
few months. They are very interested and we have offered to give them a
deposit for the machine while we test it for a week, but they don't seem to
have a machine that they want to send us. Anyone else know of another
Hello,
We created an areacode, country code, GMT offset, country code file for the
astGUIclient project last year. I believe it has all Mexican area codes in
it. If you find any errors we've love to hear about it.
http://astguiclient.sourceforge.net/phone_codes_GMT.txt
Hope this helps,
MATT---
You
should be able to get the full channel values by doing a "Action:
Command Command: Show Channels" and
picking your SIP extension out of the list it gives you of active channels. Then
you can take that and the channel that you are currently connected to, also
taken from the "Show
With a RBS(Robbed-bit) T1(in other words, not a PRI) the CallerID(Called
ANI) is sent in the digits and come across in Asterisk as part of the
extension. It is not standard, you do need to ask for it to be enabled and
you usually have to specify how you want it.
A standard way of receiving ANI on
Hello,
We tried a Dual Processor AMD system last year and were greatly
dissapointed. A single P4 system was much cheaper and actually outperformed
the Dual AMD.
Is anyone actually running an octal AMD system out there?
In our experience having more processors doesn't really matter on the x86
Hello,
We use and develop the astGUIclient suite. It is Open-source(as in GPL) and
offers Inbound and Outbound call center functions with reports, ACD,
monitoring, recording and very basic IVR scripts. Complex IVR functions need
to be custom programmed within Asterisk but that is not really that
Hello,
We've released another update to our Asterisk GUI Client suite: 1.1.0
http://astguiclient.sf.net/
Screen shots: http://astguiclient.sourceforge.net/screenshots.html
The client suite runs on both Windows and UNIX, includes the VICIDIAL
auto-dialer and is free as in GPL.
(the suite is not
The best way to figure out the manager protocols is through looking at the
manager.c source code and trial and error.
Some things just don't behave the way you think they should, some things are
not fully documented and some actions do not work in certain cercumstances
while others will.
And
Hello,
You can do either, you can send multiple Originate actions in a long line
without waiting for a response back(although the responses do usually come
back very fast) or you can open multiple connections using each one to
Originate a new call. We use the multiple connection method in the
ActionID does not return in all events related to an
Action sent, sometimes it will just send you a success message and nothing more.
Just try Originating a call from a meetme room over an outside line. You will
get about 150 lines of output and only one message will have the ActionID in it,
as possible) if the calee
phone number is ringing.
Thanks, Tom
--- mattf [EMAIL PROTECTED] wrote:
ActionID does not return in all events related to an
Action sent, sometimes
it will just send you a success message and nothing
more. Just try
Originating a call from a meetme room over an
outside line
We use astGUIclient suite, it has this functionality. Hard or soft phones
SIP, IAX or Zap
http://astguiclient.sf.net
MATT---
-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 02, 2005 4:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Hello,
The astGUiclient suite has it's own mailing list for questions like this:
https://lists.sourceforge.net/lists/listinfo/astguiclient-users
The easy fix is for you to set PHP globals to on and see if it works like
that first, also you could try making that directory writable.
MATT---
The free utility WavePad for Win32 will play and edit GSM files as well:
http://www.nch.com.au/wavepad/
To convert to/from GSM on Win32 you can use DBpowerAMP:
http://www.dbpoweramp.com/dmc.htm
And for Linux or Win32 you could use Sox of course:
http://sox.sourceforge.net/
MATT---
Hello,
A good rule of thumb for heavy perl users is to not use Fedora/RedHat. Or at
least not use rpms or the preinstalled perl on the OS. RedHat has done a lot
to screw up how perl works in the last several versions and there are a lot
of angry perl developers that have just given up on the
We are running HEAD from last night and 1.0.5 and 1.0.3 and 1.0.2 and they
all are running just fine in production environments each handling thousands
of calls a day.
I suppose reliability depends upon what you are using, but for our purposes
they all are very stable. I could do without the
Hello,
We have a call redirection system setup inhouse to send calls from an
incoming line on a T1 to an external dialed out number:
Zap(call comes in) - Asterisk - Zap(call dials out)
The call comes in on a Robbed-bit T1 (d4/ami EM Wink) and goes out on a PRI.
We are using Asterisk release
Would enabling Busydetect really help if Asterisk thinks it detects an
On-Hook?
MATT---
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Monday, February 21, 2005 7:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zap
Hello,
I have a crossover PRI(Asterisk server to PBX) and a regular telco PRI T1
line and currently have echocancel=yes and echocancelwhenbridged=yes on
those spans in zapata.conf. I was discussing CPU load with another Asterisk
user and he mentioned that PRIs don't need echo cancelation and that
Use the manager API to send a call from the meetme room to an extension that
does Monitor for a specified period of time. That is how we do it in the
astGUIclient suite and it works great.
; extensions.conf entry:
; this is used for recording conference calls, the client app sends the
filename
;
stop the recording if it is set for a period of time? Eg if
set the period as 30 minutes and the call finishes early will it cease
recording or hold up the line for 30 mins
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Wednesday, February 16
by GUI do you mean a configuration utility or a User Interface?
MATT---
-Original Message-
From: Jon Gabrielson [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 10, 2005 10:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] asterisk GUI's that
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0003490
apply the patch: app_dial_CID_nodelete.patch
and the deleting of the original callerid will stop in v1.0.5.
Also in CVS_HEAD preserving original callerid has been given a flag 'o' in
the dial string.
MATT---
-Original
mattf wrote:
Also in CVS_HEAD preserving original callerid has been given a flag 'o' in
the dial string.
I have to wonder why the default behavior was changed to this
non-standard usage though; in what situations do we want the CLID/CNAM
of the _recipient_ to be passed to them
Hello,
First, there is a mailing list for the astGUIclient suite:
https://lists.sourceforge.net/lists/listinfo/astguiclient-users
As for your problem, If you have everything set up correctly you should just
be able to run the AST_VDhopper.pl script from your Asterisk server to fill
your lead
Hello,
patching v1.0.5 on my system removed the problem for me. But yes it seems
strange that this feature was inserted into a final release with very little
documentation of the wide implications that are caused by the change.
This was corrected in CVS with the addition of a diabling flag for
, February 02, 2005 11:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: astGUIclient users should not upgrade
to Asterisk 1.0.5
Hello,
I'm not a astGUIclient user, but I'm puzzled by the following statement:
mattf [EMAIL PROTECTED] wrote
dial
out). Or if you are using Asterisk 1.0.5 simply use the patch mentioned
before to eliminate callerid altering completely.
Thanks Mark!
MATT---
-Original Message-
From: mattf [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 02, 2005 12:32 PM
To: 'Nicolás Gudiño'; 'Asterisk Users
Hello,
Just confirmed this on my end, because of the massive changes that have been
made to callerID handling in asterisk 1.0.5 many of the features of the
astGUIclient suite will not work on this new version. The latest stable
version recommended is Asterisk 1.0.3. We will work on trying to find
I'm trying to get a souped-up test machine(G5 Xserve) from Terrasoft to do
some testing in a few weeks. If/when I actually get it I'll certainly post
the results here.
In theory the G5 should mop the floor with the Intel for high-volume
Asterisk Zaptel usage, and I have heard from several
Hello,
When I talked with the VP of VOIP phone sales at Polycom about a year ago,
he was offering a dedicated engineer for the Asterisk community that would
work through issues like people have here. BUT they would ONLY do this if a
reseller came forward and committed to be the Polycom authorized
The 1.4.1 firmware and the 2.6.1 bootrom are also now on
http://www.freedomphones.net/polycom/files/
MATT---
-Original Message-
From: mattf
Sent: Wednesday, January 26, 2005 1:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: Polycom
We record meetme rooms by sending a manager Action to place a call from the
meetme room to an extension that is defined to start recording for a
predetermined amount of time, to end that recording we just send an Action
to Hangup that channel. Been working great for over a year now with over
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