Pepo wrote:
> 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.
>
I am not aware of anything directly in Asterisk, but
Am Montag, den 15.10.2007, 16:38 +0300 schrieb Cosmin Prund:
> > 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.
> > >
> >
Hi everyone, I've set up a little Asterisk system with a Digium TDM400P and
everything works splendidly except for the messages callers leave. Every
message that a caller leaves is very faint. I've already set volgain=6.0 in
voicemail.conf, and that seems better, but to be at a good volume I estima
I use vi. Not sure if it has a web interface yet.
PaulH
On Tue, 2007-10-16 at 00:51 +0200, Dovid B wrote:
> None. Asterisk vanilla is the best IMHO.
> - Original Message -
> From: Anciso, Roy
> To: asterisk-users@lists.digium.com
> Sent: Monday, Octobe
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: [ast
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]
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 t
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 sett
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} =
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 proto
> 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
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
>
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:
>>
>>
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 Ke
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
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
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 occur
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
__
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
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"
Sent: Monday, October 15, 2007 6:47 PM
Subject: Re: [asterisk
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
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
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
___
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. C
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 simple
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?
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 Ba
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 inst
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
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' com
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 co
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
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. <[
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**
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-us
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.
Do
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 D
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
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 b
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
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
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 incom
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 Intermed
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).
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 contro
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.
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 ope
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 http://www.a
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:
>
> Aster
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
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 sy
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
respon
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.
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 (Ba
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 gree
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 Provide
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 Syste
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
> timeou
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
Metro
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 s
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
> 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 faxe
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 comp
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 t
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:
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 sys
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,
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 <[EMA
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
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
> as
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 aste
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 tone
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
> > 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
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 s
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 m
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
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 ever
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 actually
X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more
information
X-ECN Telecoms-MailScanner: Found to be clean
X-ECN Telecoms-MailScanner-SpamScore: s
X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED]
X-Spam-Status: No
Brent Torrenga wrote:
> Does anyone have any tricks t
80 matches
Mail list logo