I'm sorry for sending that file. The document I sent was not ment for sending,
but just a visual aid for me, I had made a smaller txt file for sending, but I
mistakenly attached the wrong file.
My appologies for that!
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECT
Andy Deweirt wrote:
In the attachment I have drawn the architecture I want to build. Is it
possible to build such an architecture and which hardware do I need on
the dotted line to transform the digital voice back into analogue voice?
Well, first let me thank you for sending a 21KB attachment to
Title: FW: What do I still need?
Hello,
In the attachment I have drawn the architecture I want to build. Is it possible to build such an architecture and which hardware do I need on the dotted line to transform the digital voice back into analogue voice?
<>
Thanks.
Dennie
___
On Tue, 2005-02-22 at 17:00, Chris Blake wrote:
> On Tue, 2005-02-22 at 10:48, [EMAIL PROTECTED] wrote:
> > > When adding the details in AMP for when caller dials 3, I have
> > > referenced it using 'custom-myapp,s,1', and if I go to
> > > 'extensions_additional.conf' I see the following line under
On Wed, 23 Feb 2005, Rod Bacon wrote:
> No matter which version of SpanDSP I use, with which version of libtiff,
> Asterisk, ... I simply cannot send faxes.
Did you remember to add the "caller" option to txfax?
Peter
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Ken D'Ambrosio wrote:
Howdy! I'm VERY interested in your HOWTO... but the link you have,
below, times out. Any chance you could mail me the HOWTO, or point me
to a new link?
Well, linux bridging is *really* easy, here is what I have on my box
(eth0 goes to the LAN, eth1 to the netgear modem).
Howdy,
On Tue, 2005-02-22 at 17:13, beonice wrote:
> I'll let someone else speak to the missing .conf
> files.
>
> If you could post your extensions.conf
Here it is (rather long, sorry):
==
; Asterisk Management Portal (AMP)
; Copyright (C) 2004 Coalescent Sys
Hello
I am using asterisk 1.0.0, here i am facing one
problem that the email-aatchment setting for each
extesion is not working individually.
When globally attach=yes is set the voicemail will be
sent as attachment no matter for any extension if
attach=no is set for it.
Same in the case with if at
Andrew Duey wrote:
I have a * box running * version 1.0.3 with two X100P line cards in it and Cisco 7960 IP
phones. Everything seems to work pretty well with the exeption that the system hangs up
on phone calls for no apparent reason. It does this on both incoming and outgoing calls
through th
I have a * box running * version 1.0.3 with two
X100P line cards in it and Cisco 7960 IP phones. Everything seems to work
pretty well with the exeption that the system hangs up on phone calls for no
apparent reason. It does this on both incoming and outgoing calls through
the POTS line (cu
Hey Everyone -
I was going to create a visio diagram outlining how my extensions will
flow out. I was just wondering if anyone on the list may have an example
they have already done up so I can get some ideas.
Thanks
**
Richard J. Sears
Vice President
I have googled until blie in the face, WiKi'd until physically exhausted
and searched through every Asterisk repository that I can find, all to
no avail...
No matter which version of SpanDSP I use, with which version of libtiff,
Asterisk, ... I simply cannot send faxes.
I can receive faxes *pe
Dear ALL:
I find a program named "asterisk_b2bua" on
http://developer.berlios.de/projects/b2bua/
And I also download them(two components) and try to test it.
But I have not enough knowledge about asterisk. It seems a Software PBX.
Does asterisk_b2bua work? Does anybody ever try it?
I have quest
Howdy! I'm VERY interested in your HOWTO... but the link you have,
below, times out. Any chance you could mail me the HOWTO, or point me
to a new link?
Thanks much!
-Ken
[EMAIL PROTECTED] wrote:
I've created a pretty complete HOWTO on creating a Linux Bridge (using
Fedora) to shape LAN <--> W
I started working on testing [EMAIL PROTECTED] I have setup the
system with 5 phones and 1 pots line.
I am using polycom phones for this system.
Polycom's register and can make outbound calls with no
issues.
When I make an internal call... The calls go straight to vm
without ringing any phones. I
Can't find em anywhere
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paradise
Dove
Sent: Monday, February 21, 2005 9:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] sip wifi phone?
what about senao SI-780
Has anyone successfully connected a Broadvox Direct Mediatrix TA to *? I
need to try and do this for an incoming line that I have from them. They
do not have a BYOD plan and I am using them for a number I am getting
ported. It is too late to cancel the port, or else I would get
Voicepulse to take i
dean collins wrote:
Guess I'll just have to stick with running connections to the ATA's via
X100P
That's what I do here. As I have a very old plan (US$ 6/month), I only
use pkt8 to place international calls - and keep it as an emergency
backup for the US.
Check your zapata.conf that it includes the following:
busydetect=yes
busycount=3
Alex.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Dugas
Sent: Wednesday, 23 February 2005 7:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sub
Me agree too.
PaulH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Wednesday, 23 February 2005 12:37 AM
To: PHP Mechanic; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How do I do this ?
FOP
http://www.
Play around with the wink= and rxwink= options in
/etc/asterisk/zapata.conf. Try setting rxwink=200 and wink=200 and
stop and start Asterisk. It looks like asterisk is not seeing a wink
from the telco.
Do NOT use busydetect or callprogress options.
al3x * wrote:
Hello,
I've got very annoying
Please help me, i can only able to register 1 port of my 6
port fxo (sip) with asterisk, it alway last one register. not all port. how to
fix this proble.
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Marco Supino wrote:
> Hi,
>
> I tried to add the IAXTel config to my asterisk, so i can dial free
> numbers inside the US from my SIP softphone (X-lite), everything seems
> to be working, but the sound quality is terrible, the other side sounds
> like a "digitized" voice, and the voice is cut, i c
I've setup * with TDM400P w/1 FXS, 3 FXO modules.
I've one analog phone connected to TDM400P FXS module, 1 PSTN line to one of the FXO module(ZAP) , and 2 analog phones connected to Sipura 2000 (SIP).
The calls between SIPs and zap phone (fxs) are OK. But 2 issues cannot be solved:
1. To dial
Carlos Chavez wrote:
I had everything working fine until today. Today the Sipura cannot dial
anywhere. I just get the following:
Feb 10 12:48:18 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to
create/find channel
Feb 10 12:48:19 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to
I've a problem about the SIP phone registration.
1. with Linux 8.0, I can setup 2 or more Sipura-2000 boxes with no registration issue.
2. with Fedora Core 1 or Fedora Core 3, I've found that the extensions on 2nd Sipura-2000 cannot register well. This cause no dial tone from the phone connected
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rudolf
Ladyzhenskii
Sent: Tuesday, February 22, 2005 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TFTP Server
Any directory name is fine as long a
On Feb 22, 2005, at 21:15, Brian Capouch wrote:
That's for starters. I'm sure others will chime in with other evils
beyond these.
HTML mail is a favorite tool for virus writers and spammers because
it's so easy to hide nasty payloads and all those "helpful" garbage
email clients out there love
Any directory name is fine as long as you configured TFTP server to use it.
Also, from device (phone) point of view, your /TFTPBOOT directory is '/' (root)
directory on server!
Rudolf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ferguson,
Michael
Sent:
I created a different dir, "/SIPFONE"
Now I have to check if it readable by all. Thanks.
I set my Windows 2003 DHCP to assign the TFTP server's IP address,
default gateway, dns, etc, etc and the phone got all that quite well but
not picking up the files.
-Original Message-
From: [EMAIL PR
The only thing I have different in my CME dial-peers is "application
session" for each of them. Other than that, what you have works for me..
-Greg
Nathan Alberti wrote:
I have the following configuration and am obviously missing something
small that is causing * not to work as expected.
I hav
On my server (ES3) the TFTPBOOT folder is where I put my Cisco image loader
files
-Original Message-
From: Ferguson, Michael [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 22, 2005 1:25 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] TFTP Server
G'Day All,
Can anyon
On my server (ES3) the TFTPBOOT folder is where I put my Cisco image loader
files
-Original Message-
From: Ferguson, Michael [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 22, 2005 1:25 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] TFTP Server
G'Day All,
Can anyon
Hi,
setup is in /etc/xinet.d/tftp file
Default directory is /tftpboot. make sure that this directory is readable by
anyone.
Rudolf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Gary G.
Hendershot
Sent: Wednesday, February 23, 2005 9:18 AM
To: 'Asteris
The Cisco phone is defaulted to getting its IP address and related from a
DHCP server ... One of the things the phone expects to get from the DHCP
server is the address of the TFTP server ... If there is no TFTP server
address handed out by the DHCP server, the phone assumes that the DHCP
server
I'm still trying to install a HFC-s BRI card onto
[EMAIL PROTECTED] .6
I'm new to this so I probably am overlooking the
obvious.
Can I just install BRIstuff onto a fresh [EMAIL PROTECTED] install?
The BRIstuff installer downloads another * from
Digium. Will this interfere with the @home ins
On my server (ES3) the TFTPBOOT folder is where I put my Cisco image loader
files
-Original Message-
From: Ferguson, Michael [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 22, 2005 1:25 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] TFTP Server
G'Day All,
Can anyon
Title: [Asterisk-Users] TFTP Server
Check under /etc/xinetd.d/tftp. There's a server_args variable that should read like -c -s /path/to/files
This is a suse 9.1 box but should be about the same
On Tue, 2005-02-22 at 16:10, Gary G. Hendershot wrote:
On my server (ES3) the TFTPBOOT folder
-- SIP/3013-5f1c answered SIP/3000-1368
-- Attempting native bridge of SIP/3000-1368 and SIP/3013-5f1c
-- Started music on hold, class 'default', on SIP/3013-5f1c
-- Stopped music on hold on Zap/1-1
-- Stopped music on hold on SIP/3013-5f1c
-- Attempting native bridge of SIP
On my server (ES3) the TFTPBOOT folder is where I put my Cisco image loader
files
-Original Message-
From: Ferguson, Michael [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 22, 2005 1:25 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] TFTP Server
G'Day All,
Can anyon
On my server (ES3) the TFTPBOOT folder is where I put my Cisco image loader
files
-Original Message-
From: Ferguson, Michael [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 22, 2005 1:25 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] TFTP Server
G'Day All,
Can anyon
On my server (ES3) the TFTPBOOT folder is where I put my Cisco image loader
files
-Original Message-
From: Ferguson, Michael [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 22, 2005 1:25 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] TFTP Server
G'Day All,
Can anyon
On my server (ES3) the TFTPBOOT folder is where I put my Cisco image loader
files
-Original Message-
From: Ferguson, Michael [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 22, 2005 1:25 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] TFTP Server
G'Day All,
Can anyon
> exten=>2,1,Dial(capi/720:078***)
exten=>2,1,Dial(SIP/mateo01,15)
On asterisk CLI, type "show application dial"
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Awesome, thank you this will save me allot of time:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Oliver
Sent: Tuesday, February 22, 2005 2:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple Par
We use it for a client in 2 ways...
exten => _218X,1,SuperValetParking($[ ${EXTEN} + 100 ]|mylot|15|$ [${EXTEN} +
100]|10|superpark)
exten => _218X,10,Playback(vm-nobodyavail)
exten => _218X,11,Dial(SIP/${OPERATOR},15,m)
exten => _218X,12,Hangup
exten => _228X,1,SuperValetParking(${EXTEN}|mylot|1
Hey Guys
Im trying to forward a call from the asterisk mainmenue to my second
computer with X-Lite installed..
What I've done so far is this:
Installed X-lite @my win PC..
X-Lite configuration:
Menu | System Settings | SIP Proxy | default
Display Name: mateo01
User Name & Authorization User
Hello,
I am trying to set a variable using the Manager API Setvar. I am
testing with a sample php code from the wiki. But when I run it I am
getting back the error:
ERROR:
Response: Error
Message: No such channel
Do channels have different names in the manager api than they do in
the Dialplan?
Thank you any feedback would be greatly appreciated.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Tuesday, February 22, 2005 2:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple Parking Lot
From
http://www.supermicro.com/products/system/1U/6014/SYS-6014P-8R.cfm
you can see the board has the IntelR E7520 chipset.
I would suggest you note this to Digium when purchasing your TE410p, as
several people have had issues with this chipset in servers (see HP
DL380-G4), and Digium have a n
juse valetparking. I can't give you an example right now, since I'm
working on implementing it. When I'm done I'll report
On Tue, 22 Feb 2005 11:51:45 -0700, Chris Modesitt <[EMAIL PROTECTED]> wrote:
>
>
> Question: I am PBX multi-hosting several customers on one of my * servers,
> what the bes
Hi,
Hoping someone has run into the same issue.
I have an * 1.0.5 tdm400p and 2 fax machines on grandstream 486 boxes.
When a fax comes in, no problem receives it fine. When you try to send a
fax out just as the fax seems to be finishing the send you get a comms
error on the fax machine and it
How is the estimated hold time calculated?
Is this based on the average length of a call times number in queue?
Is this based on the average hold time times number in queue?
Is there some base number that is used before averages can be obtained?
Is there a way to set and/or tweak the estimated hold
Hello,
I'm trying to setup an asterisk extension to be attached to an H.323
gatekeeper so that an asterisk application (Astcc) answers H.323 calls from
any terminal logged into the gatekeeper.
I'm using asterisk's channels/h323 implementation, and I've configured the
following in h323.conf:
[genera
Warren Burstein wrote:
I'm going to be hooking FXS lines on a TDM400 to a PBX which doesn't
drop line voltage at the end of a call, so I'm going to have to use
busy detection. A few questions -
The tones are taken from the tones specified by the zone in
zaptel.conf, right? Which tones cause ha
Hi,
How can't I make my zap channel respond to incoming call without ringing. I
tried immediate=yes in zaptel.conf with no success.
Thanks.
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On 20:32, Tue 22 Feb 05, Paul A Brown wrote:
> hi All,
>
> Sorry to repost but
>
> I have a server setup that runs sip no problem. I want to try a cisco phone.
>
> how do I
>
> a) Tell if I have skinny support loaded
> b) Load it onto a debian system
>
Hi,
If you did a defaul
On Tue, 22 Feb 2005 20:11:18 -, Kevin Walsh <[EMAIL PROTECTED]> wrote:
> Umar Sear [EMAIL PROTECTED] wrote:
> > I have been using asterisk for some time now. However I have never
> > used it with any of the digium or compatable cards (Purely used for SIP).
> >
> > I understand that for using Me
hi All,
Sorry to repost but
I have a server setup that runs sip no problem. I want to try a cisco phone.
how do I
a) Tell if I have skinny support loaded
b) Load it onto a debian system
Many thanks
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Asteris
On Wed, 2005-02-23 at 01:52, Mark Eissler wrote:
> On Feb 21, 2005, at 7:35 PM, Rudolf Ladyzhenskii wrote:
>
> > Hi, all
> >
> > I am doing "prrof of concept" system. I will have two IP phones
> > connected to Asterisk box. Box itself will have 1 PSTN conenction and
> > one analog phone conencti
On Tue, February 22, 2005 10:52 am, Matt Ryanczak said:
> I saw this problem when my Sipura SPA3000 was not detecting the PSTN
> line's the CPC Signal properly. I bumped the "min CPC duration" setting
> (under PSTN line tab) from .2 to .5 and the problem went away. I have
> never had this pro
Thanks,
I thought this might be the way to do it, but I didn't want to re-invent the
proverbial wheel if one existed.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Tuesday, February 22, 2005 11:44 AM
To: 'Asterisk Users Mailing List - Non
Hi,
I tried to add the IAXTel config to my asterisk, so i can dial free
numbers inside the US from my SIP softphone (X-lite), everything seems
to be working, but the sound quality is terrible, the other side sounds
like a "digitized" voice, and the voice is cut, i cant hear a full word,
I tried
dean collins wrote:
I don't understand what the big deal about html email is?
Whats the problem with your email applications?
Why are they unable to read html?
Are the emails bigger for some reason?
There seems to be more friction about this on the list of the past week
and I just don't understand
-Original Message-
My concern is for our incoming lines. I am not sure whether to go with a
VOIP provider or to stay with our existing lines. A T-1 may also be an
option for our phone lines. We could also use the t-1 for our internet.
Does anyone know of a PSTN gateway that is fairl
Umar Sear [EMAIL PROTECTED] wrote:
> I have been using asterisk for some time now. However I have never
> used it with any of the digium or compatable cards (Purely used for SIP).
>
> I understand that for using Meetme, I need to have a timing device,
> which could either be hardware or zrdummy et
I don't understand what the big deal about html email is?
Whats the problem with your email applications?
Why are they unable to read html?
Are the emails bigger for some reason?
There seems to be more friction about this on the list of the past week
and I just don't understand it.
Dean
-
Thx Aaron, they are set as off.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron Johnson
Sent: Martes, 22 de Febrero de 2005 01:51 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MOH clicks
Anton Krall
ok, thaks for pointing that out...how can I turn off the HTML tags? I am
using a web based email client.
BTW, sorry if this has been annoying, it's not been on purpose.
Original Message
Subject: Re: [Asterisk-Users] bridging iaxtel calls to PSTN
From: "Jens Vagelpohl" <[EMAIL P
Anton Krall wrote:
Im using an analog phone connected thru a grandstream handytone 286 ata
No clues if it has silence suppresion though but seems the hickups have
become minimal now... Weird since I didn't change anything but if they show
up again Ill check the silence suppresion config.
Thx!
ok, thanks for pointing that out...
Original Message Subject: RE:
[Asterisk-Users] bridging iaxtel calls to PSTNFrom: "Rich Adamson"
<[EMAIL PROTECTED]>Date: Mon, February 21, 2005 4:00
pmTo: "Asterisk Users Mailing List - Non-Commercial
Discussion"iaxtel is
not working and hasn't
Thanks Scott.
On Tue, 22 Feb 2005 08:13:40 -0800, Scott Stingel <[EMAIL PROTECTED]> wrote:
> you may not be aware of the asterisk-biz mailing list, which is probably
> more appropriate for a discussion like this.
>
> you'll find many VoIP termination vendors hang out there too.
>
> Regards,
> S
Use account codes.
> -Original Message-
> From: Matthew Boehm [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, February 22, 2005 10:46 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Finding the true src in CDR
>
>
> Here is the setup:
>
> SIP/3044 -> SetCallerID(5551212
Write a script in your favorite scripting language to generate a .call
file which calls the user. Add it to voicemail.conf.
-Original Message-
From: Senyo Gualt-Williams [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 22, 2005 12:18 PM
To: asterisk-users@lists.digium.com
Subject: [Aste
Hello,
I've got very annoying behaviour from our asterisk PBX.
We have 12 channels T1 e&m wink start for TDM and using iax softphones
internally (iaxcomm, but tried firefly-thirdparty and discarded for
bad sound quality).
Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card.
In some cases when call
Thanks Clay Reiche.
Anyone,
Why is the 7960 looking for a call manager at 168.254.173.1?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Clay
Reiche
Sent: Tuesday, February 22, 2005 1:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subje
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 memor
Question: I am PBX multi-hosting several customers on one of
my * servers, what the best way to setup call parking to prevent company A from
picking up Company B’s parked calls ?
Any basic examples would be greatly appreciated.
Thanks
Chris.
___
hi everyone,
just a poll toknow if someone out there is using intensively
asterisk-HEAD version (mean the very last version
of asterisk).
I currently used asterisk.1.0.5 and sometimes I need to kill the process
because it's freezing (deadlock maybe, or something else...).
is this kind of problem
Edit /etc/xinetd.d/tftp
The -s argument is the root directory and make sure you set disable = no.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Tuesday, February 22, 2005 1:25 PM
To: Asterisk-Users@lists.digium.com
Subject: [Aster
Sarat Vemuri wrote:
I understand that this may be a Polycom setting and not an Asterisk setting,
but I don't know where else to ask (Polycom really doesn't support end users
.. Only resellers).
I don't think I've ever seen a SIP phone that acted this way (optional
or not). Sorry.
_
G'Day All,
Can anyone give me some direction in setting up the TFTP server on my
RadHat ES3 box?
I did quite a bit of reading, but I think I am more unsure now than
before. I found the information nebulous. TFTP is already installed. I
am trying to determine where the root directory for the tftp
On Tue, 22 Feb 2005, Jon Gabrielson wrote:
> On Tuesday 22 February 2005 07:17 am, Daniel Nyström wrote:
> > A little off-topic maybe, but it's still for the Adit used with Asterisk.
> > ;)
> >
> > I wonder where I can buy 50 pin Amphenol cables, with connector on one
> > side, and open cables on
I was wondering if Asterisk has a built in utility to have
the voicemail system dial a specific number when a voicemail is left and allow
the user to input their password to enter the voicemail system. Anyone know?
Thanks,
~Senyo
___
A
Mark Floyd wrote:
The reason I want to do this is I get a lot of calls from cell phones and
the caller ID name shows up "unavailable". Do you know if a new firmware
release is on the way?
No idea, Polycom doesn't cooperate with us and let us know about those
things.
I would suggest putting some
On 16:25, Tue 22 Feb 05, PHP Mechanic wrote:
> Hi,
>
> I wish to initate calls from a web interface, by clicking on a link and
> then connecting to the automatic outgoing call by picking up an analogue
> phone.
>
> I've got one fxs and one fxo and I wish to automate the call using a call
> fil
I will certainly look at this. However, it may not be till late this
weekend as I have a five page paper due Saturday and I have a lot of
work left on it.
This was the only thing that was keeping me from using the [EMAIL PROTECTED]
iso.
Look forward to being able to test this.
Robert
-Origi
Hello,
I've just uploaded a patch to amportal project at sourceforge, to
support Zap Extensions...
http://sourceforge.net/tracker/index.php?func=detail&aid=1146433&group_id=121515&atid=690574
I'd appreciate some feedback ;)
Greetings
Julian J. M.
On Sun, 20 Feb 2005 22:00:44 -0500, Robert
Muhammad Muzzamil Luqman wrote:
i have got a music file with extension mp3 and it is not workign with
background()
First off, why are you trying to background an MP3? If you are
backgrounding music, then you should probably be using Music on Hold.
If you are using mp3's for audio prompts in an
Hello Everyone,
I am looking into using Asterisk as our company PBX and voicemail system. I
am very familiar with Linux, but the VOIP stuff is new for me.
We are a non-proffit organization, so keeping things as cheap as possible is
very important. I am looking on some advice for best implementing
I have InAccess Networks' oh323 installed and partially working. I can call
the h.323 phone from asterisk using Dial(oh323/${IP_ADDRESS}). How do I
dial from the phone to an asterisk extension? It does not appear to me that
the phone actually registers (or attempts to register) with asterisk.
Take a look at this URL:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sound%20files
Ive used the following command
sox inputfile.wav -r 8000 -c 1 outputfile.gsm resample -ql
hope this helps
Grüsse / Best Regards
Mateo Meier
-
A standard scsi cable works great.
Just cut it in half.
Cheers,
Jon.
On Tuesday 22 February 2005 07:17 am, Daniel Nyström wrote:
> A little off-topic maybe, but it's still for the Adit used with Asterisk.
> ;)
>
> I wonder where I can buy 50 pin Amphenol cables, with connector on one
> side,
MF Hulber wrote:
I'm looking for a simple way to disable MusicOnHold in my environment.
I'm not really interested in having it and it causes too many problems
with hanging mpg123 processes and memory management errors. The problem
is, so many other modules seem to depend on it. I can't just c
On Tue, 2005-02-22 at 21:55 +0500, Muhammad Muzzamil Luqman wrote:
> i have got a music file with extension mp3 and it is not workign with
> background()
>
> is there any way to convert the mp3 to gsm or any other codec?
Yeah, we could send you to remedial linux classes where you would learn
how
I use the following recipe for this in Linux...
mp3-decoder -w outfile.wav infile.mp3
normalize outfile.wav
sox outfile.wav -r 8000 outfile.gsm
Things sound pretty good like that. You can do it with sox at one shot
but I like the normalization so all of my recordings sound approximately
the same
i have got a music file with extension mp3 and it
is not workign with background()
is there any way to convert the mp3 to gsm or any
other codec?
Kindest
Muhammad Muzzamil Luqman
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ht
Here is the setup:
SIP/3044 -> SetCallerID(5551212) -> Call out PRI
The CDR shows a src of 5551212. That is a lie! The src of that call was not
5551212, the source was 3044! The "translated source" of that call was
5551212.
How can I get "real" source of this call and not some faky nonsense?
Th
Can you not just remove the sym link to the mpg123 process so asterisk
doesn't find it therefore no music on hold?
When I was trying to get music on-hold working I had to compile and sym
link the mp123 executable - when it wasn't present I had no music on hold...
Mark
MF Hulber wrote:
I'm looki
Hi all
i am having odd problems.
nothing worng with the server starting up or anyhting like that.
asterisk server = P4 2.4 512M Ram 80gig/hd 100M/Lan Digium 4 port FXO Card
i am using X-Lite on macines on the same network all running on 100M/Lan
calling etc all works fine calling to PSTN works f
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