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Brent Torrenga wrote:
Does anyone have any tricks
Hello everyone.
I'm working on an application that needs to automatically send faxes. To
send the faxes I create .call files but the .call files mostly fail
because my lines are always congested within business hours! Is there
any trick I can use to give the end user a better chance at
Hi
Thanks for reply
Yes, there's a change. For me it's completely unacceptable, so i
reverted the patch (http://bugs.digium.com/view.php?id=10659).
For me too. This bug occur in September. Is it still present in asterisk
1.4.12.1. I also have asterisk 1.4.4 on a different box and there
Am Montag, den 15.10.2007, 11:26 +0300 schrieb Cosmin Prund:
Hello everyone.
I’m working on an application that needs to automatically send faxes.
To send the faxes I create .call files but the .call files mostly fail
because my lines are always congested within business hours! Is there
Hi,
I have a question about the combine key sequence in feature.conf.
Say, I have a featuremap for atxfer.
atxfer = *1
So I press *1 to enable atxfer. I want to know how can I adjust the
timeout second between * and 1. I found they need to be pressed
within 0.5 second to make it work. Can I
In article [EMAIL PROTECTED],
Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Oct 14, 2007 at 05:43:27AM -0700, Dominic Son wrote:
Ok, this is what worked:
EXEC System rm -rf /var/lib/asterisk/sounds/blah.gsm
the -rf eliminates the hassle.. a dream come true it worked !
-r sure wasn't
The mistake people often seem to make is to assume that
loadavg == cpu usage.
It is a good indication. Even a better indicaton to the ammount of
threads (processes) starved for CPU time.
On a quad core Linux machine it is possible to have a totally
unusable machine with a loadavg of 4 or
I think you should use a set of queues - if your skill-based requirements
are the usual suspects (speaking different languages) it's fairly easy to
set up with a master queue for each language with different priority
groups based on how good the agent is with that language. We have a good
Hello Cosmin,
it's hard to tell without first knowing what is going on on your side, but
I would not just drop call files and let Asterisk decide when to process
them - if you have hundreds of faxes pending, you risk having all lines
busy sending faxes and your other users without a dial
Hello all,
I am trying to set up asterisk and hylafax to send and receibe fax. The
machine is connected to the PSTN is an Eicon Diva BRI (1 BRI port).
My problem is that , when I send a Fax from the PSTN to this machine, the
asterisk or diva or hylafax, does not detect this call as a fax and
Hello VoipCrazy !?
On Mon, 15 Oct 2007, voip crazy wrote:
Hello all,
I am trying to set up asterisk and hylafax to send and receibe fax. The
machine is connected to the PSTN is an Eicon Diva BRI (1 BRI port).
My problem is that , when I send a Fax from the PSTN to this machine, the
On Mon, Oct 15, 2007 at 12:01:12PM +0200, Andreas Sikkema wrote:
The mistake people often seem to make is to assume that
loadavg == cpu usage.
It is a good indication. Even a better indicaton to the ammount of
threads (processes) starved for CPU time.
On a quad core Linux machine
Dear Armin,
the problem is my Eicon Diva Card does not detect aany fax-tone. Then the
call is redirect as a voice call instead a fax call.
How could I detect the fax.-tone with this kind of hardware?
How could I enable receivefax?
Thanks in advance.
VoipCrazy
2007/10/15, Armin Schindler
FRANCOIS wrote:
Hello
I am using the Asterisk version 1.2.7.1 I found that the ring time out is
set to 30s. I mean when phone A calls phone B, and the user of phone B
doesn't pick up the call in 30s , it goes on busy. How to increase this time
This really belongs on the users list,
Greetings list,
One of our asterisk boxes has been spitting out the following error this
morning:
Oct 15 12:31:50 WARNING[22300]: acl.c:306 ast_ouraddrfor: Cannot create socket
Looking at the list archives, it seems this is usually caused by insufficient
file handles on very heavily loaded
Zaheer,
this post did show up on the 11th, I am guessing few
people have attempted this, hence no feedback.
-baji.
--
On 10/11/07, Zaheer Master wrote:
Hi All,
I have done some research on Asterisk and I would like to try it in my
office. Here's what I'm looking at for my system:
Behalf Of Anselm Martin Hoffmeister wrote:
Subject: Re: [asterisk-users] About .call files when the congestion is
on myside
Am Montag, den 15.10.2007, 11:26 +0300 schrieb Cosmin Prund:
Hello everyone.
I’m working on an application that needs to automatically send faxes.
To send
On Mon, 15 Oct 2007, voip crazy wrote:
Dear Armin,
the problem is my Eicon Diva Card does not detect aany fax-tone. Then the
call is redirect as a voice call instead a fax call.
How could I detect the fax.-tone with this kind of hardware?
How could I enable receivefax?
Are we talking about
bilal ghayyad wrote:
Dear Phellepe;
? It's a bit uncommon to change other people's names.
It was 1.4 and I set priorityjumping and set
autofallthrough and look like fine, need to test more.
Ok. So you seem to have made your decision. Although I don't
understand why there's no need to do
Well, no replies to my previous post - probably too vague. I should have
said using SIP trunks.
Anyway, I have made progress. I can authenticate my Win32 system on my
Linux system, but not vice-versa. I assume this is because I run as
root in Linux, but only as a Local Administrator on my
Hi
I have setup Elastix to do some testing, and I have zoiper installed
on two machines and two ip phones(Grandstream Budge Tone-100), no
matter in what combination there is always a delay of voice between
the ends. I have set all the devices to use PCMU codec all the devices
are connected to the
I have a surplus of Digium T1/PRI cards but no FXO/FXS cards and as luck
would have it, thats what I need right now. Was wondering if anyone would
be willing to swap a Digium TDM400P with 4 FXO modules for a TE100P? I would
pay for the shipping costs.
--
Chad Whitten
Director of Operations
OK well I will try it out and see how it works!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Baji
Panchumarti
Sent: Monday, October 15, 2007 9:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk
Read the comments in features.conf
On 10/15/07, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi,
I have a question about the combine key sequence in feature.conf.
Say, I have a featuremap for atxfer.
atxfer = *1
So I press *1 to enable atxfer. I want to know how can I adjust the
timeout second
Hi all,
If I have 2 single-line SIP phones, I can still do a conference call using
Asterisk, right? For example, two people in my office are on the call, along
with 1 other person at a remote site.
Regards,
Zaheer
___
--Bandwidth and Colocation
Asterisk isn't playing my voicemail greetings even though they are defined.
Below are the relevant configs(from show dialplan) as well as the level 3
verbose messages asterisk is giving. Also a listing of the directory.
Asterisk just plays the The person at extension... message, not the
Raúl Gómez C. wrote:
Thinking about my original post, I was reluctant of installing my PBX
on a shared system, is a Dell PowerEdge 2950 with 2 Intel Xeon Dual
Core CPUs @2GHz (4 totals cores) and 4GB RAM which serves as Domain
Controller and File Server (Samba), central backup server
Dear Armin,
Bellow I send you my /etc/asterisk/capi.conf file, I just set
faxdetect=both, but the card isn`t detect an incoming fax call.
I use capicommand(receivefax|...), and work well, but I need that asterisk
or the diva card detects an incoming fax call to send it to a specific
context.
Hi,
Does anyone have any advice in how to implement PSTN failover should an
internet connection for IAX trunking go down? to route outbound to
analog lines
Can this be written into the dialplan using a GotoIf statement by
testing the whether the internet connection is up, or from a IAX/SIP
Original Message
Subject: Re:[asterisk-users] AEL2 Syntax Highlighting
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: 13/10/2007 05:24 a.m.
On Fri, Oct 12, 2007 at 05:24:29PM -0500, Perssy Llamosas wrote:
Hi,
I am looking for a syntax
On Mon, 15 Oct 2007, Robert McNaught wrote:
Does anyone have any advice in how to implement PSTN failover should an
internet connection for IAX trunking go down? to route outbound to
analog lines
Can this be written into the dialplan using a GotoIf statement by
testing the whether the
Are any of the greetings unwriteable? I'd have a situation where .WAV
would be read-only, and asterisk would overwrite the others but not that
one, and it was just coincidence that the one that was read-only was the
one that asterisk was choosing to play to me
Moj
Jeremy Mann wrote:
jamespev wrote:
We are using only SIP trunks for our provider.(we have no POTS
hardware) Is there an aggressive echo cancellation setting in this case?
No, sorry, only for Zap channels.
Moj
___
--Bandwidth and Colocation Provided by
Hello,
A few months ago, I sent an email to this list about our web conferencing
project using Ajax/IceFaces as the client.
We decided to start all over, this time using Flash as the client.
The Blindside Project aims to develop an open source webcasting and
conferencing system built on other
Sorry!
I've gotten some complaints on this; I will try this week to
mod 1.4 so that you can choose to see the single-channel unanswered
CDR's, in a new config file option. I've gotten complaints both ways,
tho, so pardon me if I get a little confused about what users out there
want from CDR's.
Hi All,
I need help with CDR issues but first let me describe the problem.
My office has 2 Asterisk PBX the first pbx is termed as the gateway
PBX (because it carries the TE card and thus Telco E1s) since all
calls are routed via this PBX. the second pbx is know as the office
pbx. this pbx
I am having a bit of a problem getting AMD to work on a new server. On
my regular office server it works like a charm. I am running Asterisk
1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and
I am using a SIP trunk to send out calls (the same one on both servers).
Just wondering what web GUI people like for asterisk. I installed
asterisk from source and I was looking at possibly installing web GUI
for system management. So far freepbx.org looks promising anybody else
have any suggestions.
Thanks
Roy Anciso
Director of Technology
Manistee
On Mon, 15 Oct 2007, voip crazy wrote:
Dear Armin,
Bellow I send you my /etc/asterisk/capi.conf file, I just set
faxdetect=both, but the card isn`t detect an incoming fax call.
I use capicommand(receivefax|...), and work well, but I need that asterisk
or the diva card detects an incoming
On Mon, 2007-10-15 at 11:42 -0500, Perssy Llamosas wrote:
Original Message
Subject: Re:[asterisk-users] AEL2 Syntax Highlighting
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: 13/10/2007 05:24 a.m.
On Fri, Oct 12, 2007 at 05:24:29PM -0500,
Matthew J. Roth wrote:
For 35 simultaneous calls, I'd recommend a dedicated server with a 3.0
GHz dual-core CPU, 2 GB of RAM, and fast SCSI disks. In my experience,
the FSB can be just as important as processor speed so keep that in mind
as you lay out your budget. You should be able to
Hi,
In the 2nd edition of the Asterisk book, there is a section recommending
running asterisk as non-root - tried this and it works. However,
asterisk does not have permissions to view certain files relating to
zaptel as in the following 'zap show status' command in the * CLI
What would be the
On Mon, Oct 15, 2007 at 10:38:09AM -0700, Robert McNaught wrote:
Hi,
In the 2nd edition of the Asterisk book, there is a section recommending
running asterisk as non-root - tried this and it works. However,
asterisk does not have permissions to view certain files relating to
zaptel as in
Try the Prescott version of the G729 .so.
That one is made for xeon's.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Lynchfield
Sent: Friday, October 12, 2007 2:36 PM
To: Asterisk Users Mailing List - Non-Commercial
Does anyone know of such a device that I can use over a network ? It would
be a pain to run a USB cable. I am thinking of devices that are like:
www.phidgets.com
http://www.smarthome.com/1132cu.html
http://www.smarthome.com/1141.html
http://www.smarthomeusa.com/Shop/wgl-irrigation//
Thanks.
Have you figured out if asterisk is crashing or not?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Scott Moseman
Sent: Friday, October 12, 2007 2:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Anciso, Roy wrote:
Just wondering what web GUI people like for asterisk. I installed
asterisk from source and I was looking at possibly installing web GUI
for system management. So far freepbx.org looks promising anybody
else have any suggestions.
Thanks
**Roy Anciso**
Director
Asterisk is not crashing. It sends back OKs to the gateway but
doesn't include any codec for the RTP, so the call gets closed. For
whatever reason, Asterisk won't talk g729 with any of my gateways, but
it will talk (and even transcode) g729 for the phones.
Scott
On 10/15/07, Power, Paul C.
I hope I am not opening a can of worms here but IMHO there is ABSOLUTELY NO
REASON TO USE SCSI anymore! For sure not for this application but most other
things too. SATA is mature now, does command queuing, and works well on 2.6
kernels. Oh, there is the issue of cost as well.
-Original
shadowym wrote:
I hope I am not opening a can of worms here but IMHO there is
ABSOLUTELY NO REASON TO USE SCSI anymore! For sure not for this
application but most other things too. SATA is mature now, does
command queuing, and works well on 2.6 kernels. Oh, there is the
issue of cost
Robert McNaught wrote:
Hi,
In the 2nd edition of the Asterisk book, there is a section recommending
running asterisk as non-root - tried this and it works. However,
asterisk does not have permissions to view certain files relating to
zaptel as in the following 'zap show status' command
At 01:58 10/14/2007, YT Lim wrote:
I don't seem to be able to find the necessary hardware
specs for an Asterisk server. What I have in mind is a
dedicated server to serve 50 or so people. All users
will use SIP phones and there will be an ISDN gateway
for outgoing/incoming calls. Do you have any
Anciso, Roy wrote:
Just wondering what web GUI people like for asterisk. I installed
asterisk from source and I was looking at possibly installing web GUI
for system management. So far freepbx.org looks promising anybody else
have any suggestions.
Thanks
Why don't you just install
On 10/15/07, Doug [EMAIL PROTECTED] wrote:
Case:
1 CodeGen 4U Server Case $80
http://tinyurl.com/bnobz
http://tinyurl.com/95s2b
http://www.newegg.com/Product/Product.aspx?Item=N82E16811182566
Or:
1 Eagle Tech ET-RMAL2025-SL Beige 2U Server Case 2 External 5.25
Drive Bays
On 10/11/07, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Yes, see the t1e1override module parameter in wct4xxp/base.c. IIRC, it's
0xff to hard code to E1 mode, and set it to 0 for T1 mode. -1 is to use
the jumper settings.
Seems like a bad design. Why not just make it a software choice??
Thanks for your suggestion, I saw mention of the asterisk-gui in a
previous post but didn't see much response on it. As I mentioned in my
original message I have installed Asterisk from source and I also have a
good understanding of how and why asterisk works. However I would like
to make it
Can I do this?
I have a x100p card on my PSTN line and I have an incoming context for
these calls which uses the s extension. I'm wanting to set up a simple
IVR and would like to be able to test the dialplan as I go. But having
to dial-in on my PSTN line each time is going to cost me money.
Hi ALL;
Any one knows a websites that has really a members
that use DUDNI wouldwide and ready to do route
exchanges?
I tried www.dundi.com but it look like still not
working, as most of its pages are not accessible
except the home page :) -
Regards
Bilal
Alan Lord wrote:
Can I do this?
I have a x100p card on my PSTN line and I have an incoming context for
these calls which uses the s extension. I'm wanting to set up a simple
IVR and would like to be able to test the dialplan as I go. But having
to dial-in on my PSTN line each time is
Yes, that will work fine Zaheer.
On 16/10/07 1:32 AM, Zaheer Master [EMAIL PROTECTED] wrote:
Hi all,
If I have 2 single-line SIP phones, I can still do a conference call using
Asterisk, right? For example, two people in my office are on the call, along
with 1 other person at a remote site.
Chanisavail does not work well for this. I would use priority jumping
(n+101).
- Original Message -
From: Alex Balashov [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Monday, October 15, 2007 6:47
None. Asterisk vanilla is the best IMHO.
- Original Message -
From: Anciso, Roy
To: asterisk-users@lists.digium.com
Sent: Monday, October 15, 2007 7:28 PM
Subject: [asterisk-users] What web GUI are people happy with?
Just wondering what web GUI people like for asterisk. I
On Tue, 16 Oct 2007, Dovid B wrote:
Chanisavail does not work well for this. I would use priority jumping
(n+101).
Why not?
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671
Dovid B wrote:
Chanisavail does not work well for this. I would use priority jumping
(n+101).
Using ChanIsAvail with the 's' option is supposed to assume a SIP
channel is occupied if it's in use ANYWHERE under asterisk's wing. For
clarification, Dovid, have your poor experiences
Thanks Matthew and every one who had replied to my post!
I will install my Sangoma A400D card on my existing server and I will give
it a try, since we have the old PBX still working (its planned to be on
operation until the end of this year) it will serve as a lab, and if there
is much trouble we
Alex Balashov wrote:
On Tue, 16 Oct 2007, Dovid B wrote:
Chanisavail does not work well for this. I would use priority jumping
(n+101).
Why not?
Priority jumping is no solution to failover, it's just an ugly
hack. ;)
I'd basically just Dial() 2 times:
Dial(SIP/...);
Dial(Zap/...);
I don't really understand how ChanIsAvail() can be of any use.
Even if it tells you that the channel is available there's no
guarantee that the call will go through.
And moreover between the ChanIsAvail() check and the Dial()
command someone else could have taken the channel.
Regards,
Philipp
At 16:13 10/15/2007, Andreas van dem Helge wrote:
On 10/15/07, Doug [EMAIL PROTECTED] wrote:
Case:
1 CodeGen 4U Server Case $80
http://tinyurl.com/bnobz
http://tinyurl.com/95s2b
http://www.newegg.com/Product/Product.aspx?Item=N82E16811182566
Or:
1 Eagle Tech
On 10/15/07, Mojo wrote:
Alan Lord wrote:
Can I do this?
I have a x100p card on my PSTN line and I have an incoming context for
these calls which uses the s extension. I'm wanting to set up a simple
IVR and would like to be able to test the dialplan as I go. But having
to dial-in
Does anyone know of such a device that I can use over a network? It would
be a pain to run a USB cable. I am thinking of devices that are like:
I think your missing the key feature of these devices, UPB/X10. UPB and X10
are communication protocols that runs across the electrical wiring in the
Quoting John Faubion [EMAIL PROTECTED]:
Does anyone know of such a device that I can use over a network? It would
be a pain to run a USB cable. I am thinking of devices that are like:
I think your missing the key feature of these devices, UPB/X10. UPB and X10
are communication protocols
On Monday 15 October 2007 19:50:03 Philipp Kempgen wrote:
I'd basically just Dial() 2 times:
Dial(SIP/...);
Dial(Zap/...);
No need for priority jumping. And not need to check if
the ChanIsAvail(). Just Dial().
Why not just do it the correct way?
Dial(SIP/,,g)
GotoIf($[${DIALSTATUS} =
On Monday 15 October 2007 17:18:00 Andreas van dem Helge wrote:
On 10/11/07, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Yes, see the t1e1override module parameter in wct4xxp/base.c. IIRC, it's
0xff to hard code to E1 mode, and set it to 0 for T1 mode. -1 is to use
the jumper settings.
On Mon, 15 Oct 2007, Jon Pounder wrote:
has anyone actually been satisfied with the performance of these
powerline signalling devices ?
yeah they make a nice cheap demo, but any time I have used them they
proved to operate randomly on their own, and not always when they were
supposed to.
Hi friends.
I am using Asterisk like voicemail of a great system with many users, How do
I can get statistics of each box in the voicemail system? something like
space, number of messages, etc.
A lot of thanks.
--
Linux User Registered #232544
Jabber : [EMAIL PROTECTED]
Whatever your many reasons, using that stuff for Asterisk is a waste of money
but go crazy if you want!
-Original Message-
From: Shaw Terwilliger [mailto:[EMAIL PROTECTED]
Sent: Monday, October 15, 2007 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
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