On Sun, Feb 04, 2007 at 08:57:28PM -0500, [EMAIL PROTECTED] wrote:
Hi everyone:
I'm trying to compile Asterisk on FreeBSD 6.0-RELEASE and I'm getting the
following error:
cc -O2 -fno-strict-aliasing -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations -Iinclude
At 19.22 04/02/2007, you wrote:
if you want exact cdr records, you must go digital.
There's still something I don't understand: when using a simple
modem on an analog line, you get correct answers from the modem:
NO ANSWER, BUSY, NO DIALTONE, etc... why is this possible
with these TDM2400
Asterisk 1.2 has no support of t.38 whatsoever, the call will drop
before t.38 is ever utilised, not even pass-thru.
1.4 Adds support for T.38 pass through only and no other sort of
faxing, the endpoint must support T.38 and you must send your call to
a T.38 gateway and you must not use NAT
On Mon, Feb 05, 2007 at 09:28:52AM +0100, Stefano Corsi wrote:
At 19.22 04/02/2007, you wrote:
if you want exact cdr records, you must go digital.
There's still something I don't understand: when using a simple
modem on an analog line, you get correct answers from the modem:
NO ANSWER,
AudioCodes is known to violate the GPL and not care at all about it.
On 12/13/06, Mike Clark [EMAIL PROTECTED] wrote:
Anyone have any experience with the Audiocodes MediaPack MP-118? We are
looking at options for a location that wishes to maintain 6 - 8 existing
analog phones in add
Hello,
Before I dig into the SIP traces maybe someone has a clue:
We have a few Snom phones, and the call-limit on the extension associated with
them is set to 1 (so the other Snom phones can blink the relevant LED).
When I call a SNOM phone (300 with the latest firmware) and the call is
Hello
I am new to Asterisk, and planning to build an internal IP PBX for the
office (100 extensions, with 8 analog lines for the outside phone lines).
What are the hardware required for this configuration other than the server?
thanks and best regards
This question has been asked many times. It really depends on what you will be
doing. Going to have transcoding, recording, rtp stream going thru the box or
not, conferencing etc. ?
- Original Message -
From: fadi mujahid
To: Asterisk Users Mailing List - Non-Commercial
Hi Friends,
I am planning to buy VoiceRD software to settingup my call centre and
planning to use Thinkbright as VoIP provider.
Anybody using the above one or two?
If you are using any of the above, please tell me your opinions.
Looking forward to your response. Thank you.
Regards,
Chandra.
Also please search the list archives. People have posted what they have used
for thier set ups and how it has worked for them.
- Original Message -
From: fadi mujahid
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, February 05, 2007 11:13 AM
Subject:
Hi list!
How to make outgoing call thru other mISDN channel group of all channels on
first group are busy?
I believe I'll need to
- Check of there is free channel on group1
- if there is free channel call thru group1
- if there are no free channels call thru group2
--
Tomislav Parčina
Lama
Hi all,
How come I occasionally get messages with the subject NOT containing
[asterisk-users] ? It stops my filters working!
Thanks,
Peter Spikings.
On Mon, 2007-02-05 at 10:06 +, Tomislav Parčina wrote:
Hi list!
How to make outgoing call thru other mISDN channel group of all channels
On Mon, Feb 05, 2007 at 10:15:39AM +, Peter Spikings wrote:
Hi all,
How come I occasionally get messages with the subject NOT containing
[asterisk-users] ? It stops my filters working!
I don't know. However there are better ways to filter than by the subject
line. Mailman injects
Hi all,
I found the following error in CLI. Anyone can tell me what is the
meaning of this error? Is it related to the codec problem? I only
allow g729 and gsm in the system. Most of the client use g729 to
connect to the server. In location A, the clients can make call
without problem.
On Mon, 2007-02-05 at 12:31 +0200, Tzafrir Cohen wrote:
On Mon, Feb 05, 2007 at 10:15:39AM +, Peter Spikings wrote:
Hi all,
How come I occasionally get messages with the subject NOT containing
[asterisk-users] ? It stops my filters working!
I don't know. However there are better
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
Sent: 05 February 2007 08:59
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Detecting answer with an analogue card
On Mon, Feb 05, 2007 at 09:28:52AM +0100,
On Sunday 04 February 2007 2:29 pm, Scott Walde wrote:
I have the following dialplan (segment) that isn't working as I expected
it to:
exten = s,n,Dial(Zap/1SIP/202SIP/203,18)
exten = s,n,Dial(Zap/1SIP/201SIP/202SIP/203,42)
The plan was to have SIP/201 added to the group of ringing phones
Hi all,
How come I occasionally get messages with the subject NOT containing
[asterisk-users] ?
I'd be interested in an answer to that one too...
It stops my filters working!
Try filtering on some of the other headers, eg:
X-BeenThere: asterisk-users@lists.digium.com
List-Id:
On Mon, 2007-02-05 at 11:06 +0100, Tomislav Parčina wrote:
Hi list!
How to make outgoing call thru other mISDN channel group of all channels on
first group are busy?
I believe I'll need to
- Check of there is free channel on group1
- if there is free channel call thru group1
- if
How does going digital help?
George
- Original Message -
From: Matteo Brancaleoni [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, February 04, 2007 5:31 PM
Subject: Re: [asterisk-users] Detecting answer with
On Mon, Feb 05, 2007 at 11:20:00AM -, Robert Jenkins wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
Sent: 05 February 2007 08:59
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Detecting
On Monday 05 February 2007 5:18 am, George Camilleri wrote:
How does going digital help?
Digital calls have a state associated with them. They can tell if the far end
actually picked up (i.e. answered) or not. Analog devices can't do this, or
rather you can make them try (progressdetect) but
I have a persistent problem with a PBX I commissioned recently. After a
few days it goes into a spasm, creating thousand of log files and giving
the message below on the CLI.
Dell PE 1600 with Sangoma A200.
pbtpbx*CLI show version
Asterisk 1.2.14 built by root @ pbtpbx.local on a i686 running
At 13.44 05/02/2007, you wrote:
On Monday 05 February 2007 5:18 am, George Camilleri wrote:
How does going digital help?
Digital calls have a state associated with them. They can tell if
the far end
actually picked up (i.e. answered) or not. Analog devices can't do this, or
rather you can
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Iirc you can put more than 1 interface in a group and it should just use
any free channel of whichever interface that has a free channel. Check
the sample config.
Hi Patrick!
Yes, I know that and I'm using that. But then I need to change
How can I play wav49 or gsm voicemail files on FC6? Nothing seems to
play them (hxplay, xine, mplayer, etc). I think I have all the normal
codec packages installed.
I can play regular wav files, but they're too big.
- Mike
___
--Bandwidth and
On Mon, Feb 05, 2007 at 08:52:52AM -0400, Chris Mason (Lists) wrote:
I have a persistent problem with a PBX I commissioned recently. After a
few days it goes into a spasm, creating thousand of log files and giving
the message below on the CLI.
Dell PE 1600 with Sangoma A200.
pbtpbx*CLI
Hi
it's possible to use a Digium TE110P Single T1 / E1 PCI Interface for supply
a E1 link to a old PABX ?
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi,
I have a problem understanding which 'h' (hangup) extension is used in
which case - It seems to vary depending on channel type. Assuming the
following simplified dialplan:
[macro-faxhere]
exten = s,1,rxfax(file)
exten = h,1,NoOp(Hangup in macro)
[fax]
exten = _X.,1,Macro(faxhere)
exten =
I haven't quite gotten this working yet but I am going to update the thread
with what I have learned. Maybe this will help the next guy who tries to
figure this out.
The trick to using the DIALSTATUS seems to be to put it in the handler for
the h (hang-up extension).
[outdialer]
Hello all,
I am looking for software for text to speech in spanish witch works with
asterisk (1.2.13).
I have tested festival and the cepstral software, both works but the quality
is so poor in the spanish language.
Someone has worked with any test to speech software with aceptable quality
in
I find this surprising. Is this fact? I don't see faxing disappearing
anytime soon. I'm surprised Asterisk/Digium would ignore it and not try
to support it. If people are to replace their old PBX's with Asterisk
faxing is almost always going to be required. We have an old POTS line
for faxing but
Hello Larry,
A quick suggestion - probably is not the best one, but maybe it will help:
exten = s,n,GotoIf($[${CALLERID} = ]?anonymous:${CALLERID}|1)
exten = s,n(anonymous),Set(CALLERID(num)=NOCID)
exten = s,n(continue),..
exten = _4XX,1,Set(CALLERID(num)=Internal
exten =
On Sun, 4 Feb 2007, Yuan LIU wrote:
I define
[globals]
myvar = ${DB(store/myvar)}
---
But when I want to use ${myvar} in the dial plan, I found that the variable
is null when Asterisk is restarted. Only a reload would force globals to
read AstDB. Other variables in globals loads fine.
Trixbox 1.2.3 - TDM400 FXOs - Flash (*) and # Not Working
Has anyone run into this problem. I cannot transfer or park a call (#) on
outgoing calls. Using Zaptel TDM400 FXO card. This may be normal but I
wanted to check.
Regards,
Juan S.
___
Try latest version of iaxLite softphone. The testing result here is that it
could work with PRI or IAX2 trunks.
2007/1/16, Patrick W. Foster [EMAIL PROTECTED]:
I have call center PCs that switch between an IBEAM SIP softphone and a
NEBU IAX softphone (for reasons
that aren't germane here).
We have several 2900s in production as VoIP servers.. no lockups.
On every server I go into the BIOS and:
* Disable USB
* Disabled uneeded things like Parallel, Serial
* Put ETH0 on a seperate IRQ from the Digium card
And everything's fine. Dell's do NOT have to share IRQs... go into your
BIOS
Hello list,
I have an application which is one large AGI. My extensions.conf
answers, calls the AGI, and hangs up the call. A second AGI is set to be
called if hangup is detected.
exten = s,1,answer
exten = s,n,agi(appmain)
exten = s,n,hangup
exten = h,1,agi(after_hangup)
The appmain AGI uses
On Mon, 5 Feb 2007, Dr. Michael J. Chudobiak wrote:
How can I play wav49 or gsm voicemail files on FC6? Nothing seems to play
them (hxplay, xine, mplayer, etc). I think I have all the normal codec
packages installed.
Have you got 'sox' installed? It comes with a command-line 'play'
On 1 Feb 2007, at 14:14, Lacy Moore - Aspendora wrote:
On 2/1/07, Andy Davidson [EMAIL PROTECTED] wrote:
What I would expect to happen, is that Asterisk would transcode
between the ulaw/alaw party, and me, wanting to listen via
g729. Is
this what *should* happen ? Worth noting that
Do you rotate Asterisk's logs with the logger or with logrotate?
Don't know. I have not modified the installation.
What do you have on logger.conf ?
[general]
; Customize the display of debug message time stamps
; this example is the ISO 8601 date format (-mm-dd HH:MM:SS)
; see
Tzafrir Cohen wrote:
Do you rotate Asterisk's logs with the logger or with logrotate?
I have never addressed this before and never seen this problem before.
The issue is causing thousand of log files to be written to the
/var/log/asterisk directory, so many that I have to use find to
Dr. Michael J. Chudobiak wrote:
How can I play wav49 or gsm voicemail files on FC6? Nothing seems to
play them (hxplay, xine, mplayer, etc). I think I have all the normal
codec packages installed.
I can play regular wav files, but they're too big.
- Mike
I have a TDM2402E with asterisk 1.4.0
Incoming calls that route back out a port have LOW volume
I am presently using an rxgain and txgain value of 4.0
What might I try different here?
I thought a saw somewhere where it wasnt recommended to go higher that 5.
How can I bring up the audio level to
I am just wondering if someone can explain the difference between
Packek2Packet Bridging vs. Native Bridging in Asterisk.
I'm basically tyring to make sure the media travels end-to-end and
I've see both of these bridging types mentioned on the asterisk
console.
Regards,
David
On Mon, 5 Feb 2007, Matt wrote:
We have several 2900s in production as VoIP servers.. no lockups.
On every server I go into the BIOS and:
* Disable USB
* Disabled uneeded things like Parallel, Serial
* Put ETH0 on a seperate IRQ from the Digium card
And everything's fine. Dell's do NOT have
Gordon Henderson wrote:
On Mon, 5 Feb 2007, Dr. Michael J. Chudobiak wrote:
How can I play wav49 or gsm voicemail files on FC6? Nothing seems to
play them (hxplay, xine, mplayer, etc). I think I have all the normal
codec packages installed.
Have you got 'sox' installed? It comes with a
Eric \ManxPower\ Wieling wrote:
There's still something I don't understand: when using a simple
modem on an analog line, you get correct answers from the modem:
NO ANSWER, BUSY, NO DIALTONE, etc... why is this possible
with these TDM2400 cards that cost twenty times as much?
I know that I'm
I've been getting a number of dropped calls today with the following
visible in the logs:
Feb 5 12:06:29 DEBUG[8340] channel.c: Bridge stops because we're zombie
or need a soft hangup: c0=SIP/peachnet-213243-b7830fd8,
c1=SIP/2142-08f4b6a0, flags: No,No,No,Yes
Feb 5 12:06:29 DEBUG[8340]
Derek Whitten wrote:
switch voicemail to .ogg format
voicemail.conf:
format=ogg
but you can't actually do that, can you?
WARNING[9933]: file.c:984 ast_writefile: No such format 'ogg'
mp3 would be better, but it doesn't work either.
WARNING[9879]: file.c:984 ast_writefile: No such format
You need to upgrade `make` to at least v3.8
On 2/1/07, Dennis Kavadas [EMAIL PROTECTED] wrote:
hi all
i'm getting the below error when trying to compile asterisk-1.4 on
redhat-9.0
any suggestions ?
make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect'
make[1]: Leaving
yep,
a nice setup though is run * as the man in the middle,
if you have a telco T or E, use a dual TE so you can inbound in one end,
do your * stuff, and forward to the legacy box in the other,
On Feb 5, 2007, at 7:29 AM, Noc Phibee wrote:
Hi
it's possible to use a Digium TE110P Single T1 /
Sorry, hadn't noticed this was already answered.
On 2/5/07, Sean Bright [EMAIL PROTECTED] wrote:
You need to upgrade `make` to at least v3.8
On 2/1/07, Dennis Kavadas [EMAIL PROTECTED] wrote:
hi all
i'm getting the below error when trying to compile asterisk-1.4 on
redhat-9.0
any
you gotta check festival site,
it works for Spanish too through external language modules,
On Feb 5, 2007, at 7:57 AM, voip crazy wrote:
Hello all,
I am looking for software for text to speech in spanish witch works
with asterisk (1.2.13).
I have tested festival and the cepstral software,
On Monday 05 February 2007 1:17 pm, Stefano Corsi wrote:
Uhm... but wait: I remember old days modem to have also a VOICE
return code... again, why should a 20$ box be able to detect VOICE
while a 2000$ card shouldn't?
Eric's wrong here; you're right. The modems of yore used inband audio
On Monday 05 February 2007 8:26 am, Stefano Corsi wrote:
Uhm... I still don't understand... Does call progress detection work
fairly well for analog cards with the US telephony system, or it's
still something experimental and randomly working? And if it's
working in US, how difficult can it be
Try latest IAX2 YakaPhone which you can get from www.yakasoftware.com.
_
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von ismail loo
Gesendet: 05 February 2007 17:16
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] IAX2
Dr. Michael J. Chudobiak wrote:
Derek Whitten wrote:
switch voicemail to .ogg format
voicemail.conf:
format=ogg
but you can't actually do that, can you?
WARNING[9933]: file.c:984 ast_writefile: No such format 'ogg'
mp3 would be better, but it doesn't work either.
WARNING[9879]:
Andrew Kohlsmith wrote:
On Monday 05 February 2007 8:26 am, Stefano Corsi wrote:
Uhm... I still don't understand... Does call progress detection work
fairly well for analog cards with the US telephony system, or it's
still something experimental and randomly working? And if it's
working in
On Monday 05 February 2007 2:58 pm, Stephen Bosch wrote:
For those of us without first-hand experience here, what happens when
using progressinband?
spontaneous hangups are the biggest one.
My own experience is that call progress detection -- mostly with respect
to remote hangup detection --
On 5 Feb 2007, at 16:28, Gordon Henderson wrote:
On Mon, 5 Feb 2007, Dr. Michael J. Chudobiak wrote:
How can I play wav49 or gsm voicemail files on FC6? Nothing seems
to play them (hxplay, xine, mplayer, etc). I think I have all the
normal codec packages installed.
Have you got 'sox'
Hi All,
i have been trying in vain to fix the missing files with Zaptel installation
for fedora 5 installation with asterisk 1.4.0. it will be great if someone
can point me a to a good website with steps in it or give me some pointers
on how to go about it.
TIA,
viopuser
I haven't quite gotten this working yet but I am going to update the
thread with what I have learned. Maybe this will help the next guy who
tries to figure this out...
The trick to using the DIALSTATUS seems to be to put it in the handler
for the h (hang-up extension).
[outdialer]
So what's the problem with connecting Asterisk to the telco with ISDN and
forgetting about analogue cards?
I think that you will get the correct records in the CDR and there should be
no problems with billing etc.
George
- Original Message -
From: Stefano Corsi [EMAIL PROTECTED]
To:
List,
Has anyone got any hints of how to setup the OpenSuSE Firewall2 with VOIP
friendly traffic shaping? The only bit I see is in the config file regarding
how to setup a simple HTB. I come from Shorewall, and am finding this
firewall to be different. Any help is appreciated.
Sincerely,
Brent
Need to deploy between 50 to 300 lightweight Linux - only browser and softphone.
Any recomendations?
-
Expecting? Get great news right away with email Auto-Check.
Try the Yahoo! Mail Beta.___
--Bandwidth and Colocation
Hi Folks,
I dont know how exactly to start... so im going to (what i think is) the
point...
In a dialplan, after i set an autohangup (with AGI) , how could i send a
sound (stream a sound ) into an open channel at X seconds before the
autohangup time get to 0 for that channel?
(Like public
On Mon, Feb 05, 2007 at 01:10:27PM -0800, A S wrote:
Hi All,
i have been trying in vain to fix the missing files with Zaptel installation
for fedora 5 installation with asterisk 1.4.0. it will be great if someone
can point me a to a good website with steps in it or give me some pointers
on
It depends on what you are trying to do, but the general answer is that
you can use the digium cards to connect Asterisk boxes to PABX's that
have E1 connections.
PaulH
On Mon, 2007-02-05 at 15:29 +0100, Noc Phibee wrote:
Hi
it's possible to use a Digium TE110P Single T1 / E1 PCI Interface
On Mon, Feb 05, 2007 at 01:46:24PM -0800, chester c young wrote:
Need to deploy between 50 to 300 lightweight Linux - only browser and
softphone.
If you want to avoid the geckos, then try kiax and konqueror, or kiax
and twinkle. It may turn out to be lighter than firefox in total.
--
On Mon, 2007-02-05 at 12:00 -0700,
[EMAIL PROTECTED] wrote:
Date: Mon, 5 Feb 2007 11:36:28 -0500
From: Andy Davidson [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Question on G.729 (was: H.264 *Not
Patented*)
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Mon, 5 Feb 2007, chester c young wrote:
Need to deploy between 50 to 300 lightweight Linux - only browser and
softphone.
Any recomendations?
Idefisk for the softphone.
Lynx for the browser ;-)
Gordon
___
--Bandwidth and Colocation provided by
On Mon, 2007-02-05 at 13:21 -0800, Michael Collins wrote:
I haven’t quite gotten this working yet but I am going to update the
thread with what I have learned. Maybe this will help the next guy who
tries to figure this out…
The trick to using the DIALSTATUS seems to be to put it in the
At 21.20 05/02/2007, you wrote:
On Monday 05 February 2007 2:58 pm, Stephen Bosch wrote:
For those of us without first-hand experience here, what happens when
using progressinband?
spontaneous hangups are the biggest one.
Ok, understood. But I'm still very curious: what is the wife test ?!
I'm trying to set up a simple test box to start developing with Asterisk.
I've got a Dell GX150 with two X100P cards. I've downloaded, printed out and
read through most of TFOT. I've also done a lot of Internet searching.
I'm getting this error:
ZT_CHANCONFIG failed on channel 1: No such device
On Monday 05 February 2007 5:50 pm, Stefano Corsi wrote:
Ok, understood. But I'm still very curious: what is the wife test ?! :)
It doesn't matter how many bells and whistles it has, it won't stay if it
can't let her place a simple, straightforward phone call and act just like
any other normal
On Mon, Feb 05, 2007 at 05:51:44PM -0500, David Ruggles wrote:
I'm trying to set up a simple test box to start developing with Asterisk.
I've got a Dell GX150 with two X100P cards. I've downloaded, printed out and
read through most of TFOT. I've also done a lot of Internet searching.
I'm
Hi,
I'm looking for a mean to send digital data over
an E1 line, just like isdn4linux or Capi via AVM's FritzCard
is able to do it with BRI lines (e.g. for PPP or ISDN raw
connections).
I'm not looking for modulated audio data representing
digital data, like fax or the analogue modems of former
At 10:17 AM 2/5/2007, you wrote:
Uhm... but wait: I remember old days modem to have also a VOICE
return code... again, why should a 20$ box be able to detect VOICE
while a 2000$ card shouldn't?
My guess is answered followed by no modem tone equaled VOICE.
Ira
I have Meetme conferencing compiled for Debian as per
http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the
outgoing dir, and it intitiates a call to a local extension as a
channel, but the call seems to block on the Meetme() command. That
extension completes the outgoing
point to point E1 lines? Or are you interfacing to a PSTN network for
local calling/receiving?
PTP E1
http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration
On 2/5/07, Roger Schreiter [EMAIL PROTECTED] wrote:
Hi,
I'm looking for a mean to send digital data over
an E1 line, just like
Shane Spencer schrieb:
point to point E1 lines? Or are you interfacing to a PSTN network for
local calling/receiving?
Hi,
yes, PSTN. Normal operation is ordinary voice.
Hm, the hybrid configuration mentioned in your link
may serve as a workaround anyway. I should read this further.
I did a NoOp and see what the callerid was and when coming from the SIP
Ext-Voip it is set to the Extension Number of the SIP Extension (as you would
expect).
When coming from the Panasonic the CallerID is blank, I tried setting it to
nothing again, and I tried setting it to the callerid of
Hey All,
I'll be configuring an asterisk box to be the voicemail server to an old
Merlin system which had an octel 100 voicemail server that is now dying.
My question is simple: do I need to stick an FXO card in the asterisk
box? My logic is that if the Merlin Magix system is actually
David Ruggles wrote:
I'm trying to set up a simple test box to start developing with Asterisk.
I've got a Dell GX150 with two X100P cards. I've downloaded, printed out and
read through most of TFOT. I've also done a lot of Internet searching.
I'm getting this error:
ZT_CHANCONFIG failed on
Will the Asterisk box be hooked up to external lines on the Merlin, or
extension lines?
External - FXS
Extension - FXO
later,
PaulH
On Mon, 2007-02-05 at 20:03 -0500, Jeronimo Romero wrote:
Hey All,
I’ll be configuring an asterisk box to be the voicemail server to an
old Merlin
FXS cards generate ring (you connect a station to it and it rings).
FXO cards sink ring (they take ring from the office).
If the Octel needs ring (which it most likely does), you would need an FXS
card to generate ring for it to answer. An FXO would take ring from the
vmail server, which, in
Andrew D Kirch wrote:
David Ruggles wrote:
I'm trying to set up a simple test box to start developing with Asterisk.
I've got a Dell GX150 with two X100P cards. I've downloaded, printed
out and
read through most of TFOT. I've also done a lot of Internet searching.
I'm getting this error:
On Mon, 2007-02-05 at 22:37 +, Gordon Henderson wrote:
On Mon, 5 Feb 2007, chester c young wrote:
Need to deploy between 50 to 300 lightweight Linux - only browser and
softphone.
Any recomendations?
Idefisk for the softphone.
I agree idefisk. Is light and supports IAX2.
Try to execute the command: lspci
if the card is ok, you will found some informations such as:
01:00.0 Communication controller: Motorola: Unknown device 5608
or
something likes tigerjet
If you can't find them, I think you have to unplug the card and insert it to
try www.unifycall.com . they provide IAX2 termination for VOIP beginners and
individual end-users. No minimum usage.
2007/2/3, Vicky [EMAIL PROTECTED]:
Voipjet locks $1.2 per running call and unlocks when call ends .. so $12 =
10 simultaneous calls ( if rate is 1.2 cents ) .
On 02/02/07,
I had a clone X100P (ordinary modem with Motorola chipsets) It works fine
for me. My 2 cents: If you are just using that to learn the basics of
Asterisk yes go on and continue, but for production purpose you should go
with the TDM400P card from Digium.
Anyway I did not encounter any problem with
Try just one card to begin with
Follow all the other suggestions regarding seeing if Linux can see the card
Disable all unused ports, such as serial and parallel, and USB
Don't waste your money on the TDM400
IF you want a serious analog card, go with the Sangoma A200. 5 year
warranty, real
Steve Davies wrote:
I have a problem understanding which 'h' (hangup) extension is used in
which case - It seems to vary depending on channel type.
It doesn't. It depends on which side of the call hangs up. h is
executed when the callER hangs up.
If you want to handle instances of the
Matthew Rubenstein wrote:
The real advantage in choosing an AGI (or CGI or ...) platform/language
is *reusing* the existing code that already runs on that platform, with
Well of course you should pick whatever AGI implementation matches the
rest of your environment best.
minimal
How can I access an environmental variable in Asterisk 1.2.5?
It should be possible according to:
http://www.voip-info.org/wiki/view/Asterisk+variables
which says:
Environment Variables
You may access unix environment variables using the syntax:
${ENV(foo)}
${ENV(ASTERISK_PROMPT)}: the
i couldnt agree more with Brian,
i'm sure we'll see more improvement in code and more improvement in asterisk
business edition.
Al
=
I was wondering when this would happen. A lot of successful and prospering
open source company like yours seems to do this.
Much like Google
I am trying to get called back with a DISA dial tone when I call a trigger
number. I got it to work almost the way I want, this is the callback
context:
[callback]
exten= 501,1,Congestion()
exten= 501,2,Hangup()
exten =h,1,System(cp /etc/asterisk/callback.info
/var/spool/asterisk/outgoing)
I've a problem with inserting a pause and dialing additional numbers
when going through Sipura-3000
exten = _12,1,Dial(SIP/[EMAIL PROTECTED],30,D(ww18))
D() doesn't work as it sends the DTMF tones right after FXS connects
to FXO; though, I want insert a pause and send additional numbers
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