Mike Lynchfield a écrit :
try not using dst.. maybe its a regex on te fieldname that matches for
reserved keywords..
try pre_dest instead
On 2/28/07, Bayrouni [EMAIL PROTECTED] wrote:
Hello,
I created a new field named pre_dst of type varchar(80) exactly like dst
field in cdr table.
Edgar Luna a écrit :
Hi,
On Wed, 2007-02-28 at 23:43 +0100, Bayrouni wrote:
Hello,
In the dialplan I put:
exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})
and when I call, all goes fine except that pre_dst has always NULL value
in cdr.
Do you know why?
Is something wrong I did?
As far
Hi all!
Working on the following brain-scratcher. I am setting up a Trixbox
system for someone who uses 'provider A'. Everything works fine, except
for the IVR: keypresses by callers are not being detected.
Just for testing I added my own provider, 'provider B' to their system.
And then the IVR
Hi,
I will like to know if anyone would guide me about how I can to interconnect
one SIEMENS HiPATH 3700 with Asterisk.
HiPATH have VoIP card and my idea is to do one un IP trunk between them so
we would to transfer calls and services (voicemail, IVR,..) between both.
We havent PRI ports unused
From: Yuan LIU [EMAIL PROTECTED]
Date: Wed, 28 Feb 2007 21:24:56 -0800
From: kjcsb [EMAIL PROTECTED]
Date: Wed, 28 Feb 2007 18:23:46 -0800 (PST)
Check out /path/to/src/asterisk/doc/README.variables
${DIALEDPEERNUMBER} would give it to me if I sliced it up.
exten =
Hello!
I'm using asterisk 1.2.13, in extension.ael is set Dial(SIP/${EXTEN},12,tT)
but only calling party can do forward. How to configure '*' to take this
possibility to called party?
ps
both calling/called use sip
--
___
--Bandwidth and Colocation
On Wed, 2007-02-28 at 12:14 -0500, Steve Totaro wrote:
Iban Lopetegi Zinkunegi wrote:
28th February
I am working with Asterisk 1.2.15. I have configured sip.conf for two
soft phones (I am using Xlite).I have installed the Bluez stack and so
far, i manage to make a phone call from a
**All,
**
**I'm guessing no one knows the answer as to why when I register with a
**VSP I am not sending a Port number with the registration but only my IP
**address. If anyone has any answers it would be greatly appreciated.
**
**From: [EMAIL PROTECTED]
**[mailto:[EMAIL PROTECTED] On Behalf Of
So does asterisk (Albeit with a commercial package)
http://www.attractel.com/t38.html
Lee Howard wrote:
Matt Riddell [NZ] wrote:
Does OpenPBX do a T.38 gateway then?
Yes, it does.
Lee.
___
--Bandwidth and Colocation provided by Easynews.com --
Eric ManxPower Wieling wrote:
Asterisk gets very upset when DNS is down. You might want to confirm
that /etc/hosts has entries for ALL interfaces in that system. That
should cause the system to not issue a DNS request to resolve local IPs.
Asterisk also seems to barf if it makes a
Only Asterisk 1.4.0 has chan_sip.c with T.38 code in it. Although it
still doesn't work because it's full of bugs. Seems to me that
developers just pasted the T.38 patch code from the branch developing
that issue, and nothing else have done to improve it. It has to be debugged.
Regards,
nik600 wrote:
actually it isnìt released under any type of licence.
if you want i can put the code on my web site
(but no earlier than the next week)
Please do. And it wouldn't hurt if you, somewhere on the page, put that
is released under GPL or something similar.
--
Tomislav Parcina
Olle E Johansson wrote:
However, the 1.4.0 release
is buggy, so either use 1.4 from subversion or wait for 1.4.1.
Have you put this information somewhere on web page of Asterisk? I think
its fair enough to say - look, this doesn't work as it should, use 1.2X
or 1.4 from subversion.
--
James Fromm wrote:
I've reviewed the bugs reports. I didn't see anything that applied to
this. Have you? Could you point it out to me?
Just for the record, I believe this is what you are looking for.
http://bugs.digium.com/view.php?id=8800
--
Tomislav Parcina
[EMAIL PROTECTED]
Hi,
fyi, I use Asterisk 1.2.9.1
In some scenarios, we receive call from PSTN without Callerd ID Name (which
is normal).
I would like to transfer this call to another softswitch. Again, I would
like to let this this CallerID Name Empty.
If I look at the logs, I can see
-- Executing
Hi
when I turn on my PC I able to load the drivers and start my card,
if I reboot the PC I have the following error
ztcfg -vvv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel
Has anyone used this billing application with Asterisk? I have one
potential costumer (hotel) that will use application that connects with
Fidelio/Micros and so they can use Revolution Call Accounting Desktop
for billing.
More info about product you can find on this page
Hello all
i have an asterisk setup integrated with mysql via odbc driver
myproblem is:
when i reading my voicemails, in advanced options the following functions
not working with realtime asterisk but working with flat files.
1. Reply to the message(option no:1)
2.Leave a message(option no:5)
Hi,
Now that I have Asterisk up and running I would like to find a good SIP gateway
in the UK. I have looked at sipgate.co.uk and they look pretty reasonable. I
am looking for peoples recommendations.
Apologies if this is the incorrect forum for this type of request.
Regards,
--
--[ UxBoD
On Thu, Mar 01, 2007 at 06:21:38AM -0500, Il Neofita wrote:
Hi
when I turn on my PC I able to load the drivers and start my card,
if I reboot the PC I have the following error
ztcfg -vvv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default)
I have used www.voiptalk.org for a number of years with their IAX2
connectivity and they seem very reliable with no echo issues. They will also
change the CID to your number if you fax them proof of ownership.
Chris
- Original Message -
From: --[ UxBoD ]-- [EMAIL PROTECTED]
To:
I'm looking for a recipe for a 3 way call where one of the parties can
(without using the flash button) dial-out and add a third participant to
the call. I tried Googling but it seems I'm missing a key search term.
The reason I wanted to avoid using the flash button is that some
handsets don't
On Thu, Mar 01, 2007 at 12:17:49PM -, Chris Stenton wrote:
I have used www.voiptalk.org for a number of years with their IAX2
connectivity and they seem very reliable with no echo issues. They will
also change the CID to your number if you fax them proof of ownership.
There's several
I have used www.voiptalk.org for a number of years with their IAX2
connectivity and they seem very reliable with no echo issues. They will also
Second that. Not cheap but reliable and been there for years.
___
--Bandwidth and Colocation provided by
Yes asterisk was stopped and restarted. ztcfg was not rerun. I've never
had to rerun that when I made changes in that file before, but we can try
it.
MY WISH: TelCo Switchmen could talk intelligently about the protocols used
on PRIs!
On 2/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
Matt
Hello,
We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to
the telco, users mainly use snom 320/300 SIP phones.
When dialing to an external phone number with custom ringback music, users
reported that they could not hear the music but can only hear the standard
ring tone
Dave Cotton wrote:
On Wed, 2007-02-28 at 12:14 -0500, Steve Totaro wrote:
Iban Lopetegi Zinkunegi wrote:
28th February
I am working with Asterisk 1.2.15. I have configured sip.conf for two
soft phones (I am using Xlite).I have installed the Bluez stack and so
far, i manage to make a
Vincent Tam wrote:
Hello,
We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect
to the telco, users mainly use snom 320/300 SIP phones.
When dialing to an external phone number with custom ringback music,
users reported that they could not hear the music but can only hear
the
Vincent Tam wrote:
Hello,
We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect
to the telco, users mainly use snom 320/300 SIP phones.
When dialing to an external phone number with custom ringback music,
users reported that they could not hear the music but can only hear
Hi guys,
I am planning to connect two Asterisk boxes that are currently running
in two different countries, using IAX.
I was wondering if anyone could provide me with some links or suggestion
regarding best practices in connecting two Asterisk in such way. I guess
many of you have already tried
Peter Gradwell wrote:
mmm, but as you've seen, some customers like using multiple codecs. The
cisco kit is able to support a raft of options - and it does transcoding
very nicely - so the optimum solution is to have the cisco + customer's
asterisk agree on the same codec, and then have our
I believe I noticed that I had upgraded the kernel, but not yet restarted.
I restarted, and I think that was all I had to do to get it running again.
--Mike
Message: 14
Date: Wed, 21 Feb 2007 18:01:22 +0200
From: Tzafrir Cohen [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Zaptel 1.4.0
To:
Asterisk wrote:
Hi guys,
I am planning to connect two Asterisk boxes that are currently running
in two different countries, using IAX.
I was wondering if anyone could provide me with some links or suggestion
regarding best practices in connecting two Asterisk in such way. I guess
many of you
Mike,
Did you tried with make all instead of make linux26? That worked for me on
FC5.
On FC6, I have to reinstall everything and worked with make all.
Carlos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett
Sent: Thursday, March 01, 2007
On Thu, 01 Mar 2007 17:29:57 +1300
Matt Riddell (NZ) [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mike Lynchfield wrote:
We have decided to allow our tech's to do support for
non-clients of voicemeup.com
This should normally be kept on the Asterisk-Biz list
It must be the challenge response bug. They are still using 1.0.x and I
turned off challenge response on the phone. I made the change last week,
but I haven't heard from the user one way or the other. This server is
slated to be upgraded to 1.4.0.
--Mike
--
hi eveybody,
after many test with your help and the irc channels help, i get the
led on TE110P green
with this config:
span=1,1,0,ccs,ami
= alarms OK Green Led
but the provider say that i have to set my span to this
span=1,1,0,ccs,hdb3,crc4
= alarms: YEL/RED
i can't make call's yet
On 3/1/07, Tomislav Parcina [EMAIL PROTECTED] wrote:
nik600 wrote:
actually it isnìt released under any type of licence.
if you want i can put the code on my web site
(but no earlier than the next week)
Please do. And it wouldn't hurt if you, somewhere on the page, put that
is released under
Try this:
/etc/init.d/zaptel start
Than do lsmod |grep zaptel and it should show zaptel loaded
Ricardo.
Mike Hammett wrote:
I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make
install and I don’t see any errors. This is out of my modprobe.conf:
install tor2
On Thu, 1 Mar 2007, --[ UxBoD ]-- wrote:
Hi,
Now that I have Asterisk up and running I would like to find a good SIP
gateway in the UK. I have looked at sipgate.co.uk and they look pretty
reasonable. I am looking for peoples recommendations.
There are dozens in the UK. Sipgate is a
Hi all,
I'm trying to use SIPP (http://sipp.sourceforge.net/) to stress-test our
asterisk installation. We have a very simple dialplan that uses FastAgi.
I'm finding that all calls to GET VARIABLE from the FastAgi are
returning null when the dialplan is invoked from sipp -- and they work
This should be easy, but I can't find the right wildcard.
Right now I have
exten = _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,wW)
for international and for local
exten = _9NXX,n,Dial(${TRUNK}/${EXTEN:1},,wW)
The problem is if the call isn't typed in, then you press dial, we have
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Trevor Peirce
Sent: Wednesday, February 28, 2007 21:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help: CallerID Name not being sent on
Not yet I'll pay for you help. I have been to the sineapps page many
times.I know you guys got skillz
On 2/28/07, Matt Riddell (NZ) [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Supa wrote:
I using my provdier like so SIP/Telasip-gw4/5198843344 when
Hi Guys,
I am a novice of Asterisk and I need some experts help to understand what I
can get out of it.
I need to make multiple calls (let say 50) at once to autoanswering
softphones on a LAN and send all of them the same message that they will
repeat with loudspeakers in the same environment.
I
I pick up the handset and get a dialtone. I press 9011331234567 or
something international. Before I can finish, the local
option kicks in
because it saw 9.
Is there a way to say
_9[2-9]NXXX or something like that?
Are you sure the handset is not processing that call string,
I've been using Voiptalk.org for about a year now and it passes the wife
test no problem at all. IAX2 is supported with trunking to save a bit of
bandwidth.
I use sipgate for an incomming ringback number very very handy.
I use Gradwell for domain reg etc and they are excellent I've no reason to
Hi for all
I have one queue and one agent. That agent has a SPA941 with 3 lines
configured to an asterisk. That agent logins into queue.
If two clients call into that queue and the agent receive the two calls (one
for any different line). It is possible that exist any configuration on
asterisk
Could someone provide some steps for troubleshooting Realtime? I can't see
any signs that it's working. I followed and double-checked a few different
guides around the net, but haven't been able to figure it out.
___
--Bandwidth and
I wouldn't do that with softphones, unless the softphones are designed
to do this.
The delay will vary depending on the audio card, OS, and drivers.
(And the phones might not all answer at the same time, but if you use
music on hold or so to play that should not be a problem).
[EMAIL
-- Forwarded message --
From: Benito Camelas [EMAIL PROTECTED]
Date: Wed, 28 Feb 2007 11:21:52 +0100
Subject: Problem with TE212P
To: asterisk-users@lists.digium.com
Hello.
I have a TE212 configured in E1 mode.
This is shown in a cat /proc/zaptel/2 and 3 (where the card is
Yeh, I'm sure. If I watch the debug logs in *, I see each digit running
checks to see if it matches a dialplan yet. :)
Rob
McGhee, Stefano wrote:
I pick up the handset and get a dialtone. I press 9011331234567 or
something international. Before I can finish, the local
option kicks in
[EMAIL PROTECTED] wrote:
Hi Guys,
I am a novice of Asterisk and I need some experts help to understand
what I can get out of it.
I need to make multiple calls (let say 50) at once to autoanswering
softphones on a LAN and send all of them the same message that they
will repeat with
Hi,
if no cable is connected you'll get the red alarm
can you tell me the schema of your corssover cable.
try yo connect the 2 slots berween them and make internal calls(i 'm
not sure of this option but somemone called shimi has asked me if i
have 2 TE110P to test if my cable my config are ok)
i
Iban,
For me, it seems to be the codec. Which one are you using?
On 3/1/07, Steve Totaro [EMAIL PROTECTED] wrote:
Dave Cotton wrote:
On Wed, 2007-02-28 at 12:14 -0500, Steve Totaro wrote:
Iban Lopetegi Zinkunegi wrote:
28th February
I am working with Asterisk 1.2.15. I have configured
Mike Hammett wrote:
Could someone provide some steps for troubleshooting Realtime? I can’t
see any signs that it’s working. I followed and double-checked a few
different guides around the net, but haven’t been able to figure it out.
You don't say which version you're running.
I *think* the
All the calls you listed in your example were simultaneous calls *from
the same user*. Is this what you were intending?
I'm assuming this 'cause the Peer column contains the same IP address
each time, and the User column contains the same User number. Only the
Call ID changes.
Moreover,
I've gotten hold of a Tesco Internet Phone which is a dect phone with
the base connecting to the pc via usb.
Has anyone been able to get this working with any softphone like xlite ?
It seems as if the tesco internet phone uses IAX - the software that
comes with it is a rebranded firefly (or
What do you mean by codec? i am using the release posted by Theo in
http://crazygreek.co.uk/content/chan_bluetooth.
From: Facundo Ameal [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List -
The phones are provide by Freshtel in Australia if that's any help.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
i'm sorry but due to some problem the software will be released not
first than Wednesday 7/02/2007. i'll post a message .
bye
On 3/1/07, nik600 [EMAIL PROTECTED] wrote:
On 3/1/07, Tomislav Parcina [EMAIL PROTECTED] wrote:
nik600 wrote:
actually it isnìt released under any type of licence.
Yeah, that's where firefly comes from, doesn't it.
I've got the base station plugged in, and the handset connected to it,
but it always says pc unavailable.
My system (xp) sees a usb phone for speakers and microphone, but I
can't get it to work.
Julian.
Dean Collins wrote:
The phones are
Brian Capouch wrote:
Mike Hammett wrote:
Could someone provide some steps for troubleshooting Realtime? I can't
see any signs that it's working. I followed and double-checked a few
different guides around the net, but haven't been able to figure it out.
You don't say which version
plug it in a linux box and tell us what it is please, generic-usb-audio
or what?
Bails
Julian Lyndon-Smith wrote:
Yeah, that's where firefly comes from, doesn't it.
I've got the base station plugged in, and the handset connected to it,
but it always says pc unavailable.
My system (xp)
queue show Show status of a specified queue
realtime load Used to print out RealTime variables.
realtime update Used to update RealTime variables.
restart gracefully Restart Asterisk gracefully
Aiur*CLI realtime load
You must supply a family name, a
Disregard lat post, suddenly saw DECT, what is the output of lsusb though?
Julian Lyndon-Smith wrote:
Yeah, that's where firefly comes from, doesn't it.
I've got the base station plugged in, and the handset connected to it,
but it always says pc unavailable.
My system (xp) sees a usb phone
What do you have setup in the res_mysql.conf file and extconfig.conf files?
Have you installed the asterisk addons for 1.4 to get support for mysql?
On 3/1/07, Mike Hammett [EMAIL PROTECTED] wrote:
queue show Show status of a specified queue
realtime load Used to
C F wrote:
If the temp message exists then that will play. The user has to log
into the mailbox (app_voicemailmain) and select 0 for mailbox options,
and delete the temp message. Or you could do it using the shell.
I finally resolved this problem by putting format=wav in the general
section of
Any idea when 1.4.1 will be available. There is a bug fix in the cvs
head that I need, and I don't want to run the cvs build on a
production machine.
Thanks...
--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
___
--Bandwidth and Colocation
On Thu, 2007-03-01 at 16:39 -0500, Forrest Beck wrote:
Any idea when 1.4.1 will be available. There is a bug fix in the cvs
head that I need, and I don't want to run the cvs build on a
production machine.
Thanks...
I understand perfectly what you meant, but just a reminder that asterisk
Forrest Beck wrote:
Any idea when 1.4.1 will be available. There is a bug fix in the cvs
head that I need, and I don't want to run the cvs build on a
production machine.
Thanks...
I wouldn't be building anything at all on a production machine without
doing some testing on another machine
Hi everyone,
I am trying to get Asterisk 1.4 running on CentOS 4.4 (Kernel
2.6.9-42.0.10.ELsmp) and am having a lot of trouble getting asterisk
running on it. I had a fair bit of success with the ATrpms binaries
(Zaptel worked but asterisk failed to startup because it couldn't find
the speex
On Fri, Mar 02, 2007 at 10:05:54AM +1100, Devraj Mukherjee wrote:
Hi everyone,
I am trying to get Asterisk 1.4 running on CentOS 4.4 (Kernel
2.6.9-42.0.10.ELsmp) and am having a lot of trouble getting asterisk
running on it. I had a fair bit of success with the ATrpms binaries
(Zaptel
Thanks for saving me the time. I will try and yum from ATrpms.
On 3/2/07, Axel Thimm [EMAIL PROTECTED] wrote:
On Fri, Mar 02, 2007 at 10:05:54AM +1100, Devraj Mukherjee wrote:
Hi everyone,
I am trying to get Asterisk 1.4 running on CentOS 4.4 (Kernel
2.6.9-42.0.10.ELsmp) and am having a lot
How to configure Digium S101i adapter to work with pickupexten *8 ?
I have few Sipura adapters and *8 work OK but my new Digium S101i
refuses to cooperate.
--
#Joseph
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
Has anyone managed to get gtalktovoip working at all? If so please
explain.
http://www.gtalk2voip.com/faq.shtml
2. Q: Ok, how can I call Google Talk, MSN or Yahoo users from SIP ?
A: This is a major feature of our gateway and it is very easy.
oGTalk: [EMAIL PROTECTED] can be
I have users in my dialplan that go from SIP to Cell
When they are at their desk and they hit reject call, it goes to the
next thing in the dialplan, thus transferring to their cell. Not what
they want. Is it possible to change the reject button to make it go to
voice mail or a new ext?
I have a SIP user and a remote IAX device
I want both to ring 3 times then if neiter pick up it to go to the next
thing in the dialplan. Can you do this from the dialplan or do I need
to set a hunt group up?
Thanks
Jason
___
--Bandwidth and
From: Jason Walker [EMAIL PROTECTED]
Date: Thu, 01 Mar 2007 18:06:46 -0600
I have a SIP user and a remote IAX device
I want both to ring 3 times then if neiter pick up it to go to the next
thing in the dialplan. Can you do this from the dialplan or do I need to
set a hunt group up?
Thanks
Thank you for the answer
after
modprobe wctdm
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
FATAL: Error running install command for wctdm
/proc/zaptel/ (empty)
/usr/src/asterisk/zaptel-1.2.14/xpp/utils/genzaptelconf -l (no result)
On 3/1/07, Tzafrir Cohen [EMAIL
On 2/28/07, Webster, Andrew [EMAIL PROTECTED] wrote:
Outbound calls on my Telus PRI aren't taking the Name portion of the
callerID. I've looked at the logs, and it is being set (see below), but the
PRI debug output doesn't show the name being sent anywhere. As a result,
received calls always
You sure that you have local service on the PRI? Maybe it's just an LD PRI?
On 3/1/07, Matt [EMAIL PROTECTED] wrote:
Yes asterisk was stopped and restarted. ztcfg was not rerun. I've never
had to rerun that when I made changes in that file before, but we can try
it.
MY WISH: TelCo
I have users in my dialplan that go from SIP to Cell
When they are at their desk and they hit reject call, it goes to the
next thing in the dialplan, thus transferring to their cell. Not what
they want. Is it possible to change the reject button to make it go to
voice mail or a new ext?
I
What do you mean by outbound CallerID Name? So that when calling a
POTS with CallerID service from telco the Name should show up as you
send it?
If the answer to the above is yes, then stop trying to do that. It
won't work, as the name that the POTS subscriber sees is NOT the one
you send, but
Hi,All,
I wang to use AGI in asterisk1.4.
AGI file / myperl.agi
#!/usr/bin/perl
use strict;
..
print STDERR 7. Testing GET VARIABLE...;
print GET VARIABLE EXTEN \\\n;
my $result = STDIN;
checkresult($result);
..
when the agi execute; asterisk
李君 wrote:
Hi,All,
I wang to use AGI in asterisk1.4.
AGI file / myperl.agi
#!/usr/bin/perl
use strict;
..
print STDERR 7. Testing GET VARIABLE...;
print GET VARIABLE EXTEN \\\n;
my $result = STDIN;
checkresult($result);
..
I don't know perl,
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi Axel,
Everything installed and working well. Thanks very much. Quick
question, do you have MySQL support compiled into the rpms?
On 3/2/07, Axel Thimm [EMAIL PROTECTED] wrote:
On Fri, Mar 02, 2007 at 10:05:54AM +1100, Devraj Mukherjee wrote:
Hi everyone,
I am trying to get Asterisk 1.4
I'm still stuck on just exactly where in my extensions.conf file I
should put the code below. I'm running 1.2.14 of asterisk.
Chris Griffin
[EMAIL PROTECTED]
On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:
I've installed Sven Slezak's Notify module. He gives the follow as an
example line
From: Àî¾ý [EMAIL PROTECTED]
Date: Fri, 2 Mar 2007 10:53:04 +0800
Hi,All,
I wang to use AGI in asterisk1.4.
AGI file / myperl.agi
#!/usr/bin/perl
use strict;
..
print STDERR 7. Testing GET VARIABLE...;
print GET VARIABLE EXTEN \\\n;
Why do you want put after variable
I'm still stuck on just exactly where in my extensions.conf file I
should put the code below.
Chris Griffin
[EMAIL PROTECTED]
On Feb 28, 2007, at 9:55 PM, Patrick wrote:
On Wed, 2007-02-28 at 23:28 -0600, voiplist wrote:
Thanks for the link..
As for Google, I know how to use it. I
I've been using voiptalk.org for about two years now and have never had
any problems. I've been using them for my outgoing business calls for a
year and am starting to use them for some incoming calls, which is some
indication of my comfortableness with their service.
My reluctance to move
On Fri, 02 Mar 2007 07:26:03 +, George Gardiner [EMAIL PROTECTED] wrote:
I've been using voiptalk.org for about two years now and have never had
any problems. I've been using them for my outgoing business calls for a
year and am starting to use them for some incoming calls, which is some
On Thursday 01 March 2007 20:33, bails wrote:
plug it in a linux box and tell us what it is please,
generic-usb-audio or what?
Bails
Julian Lyndon-Smith wrote:
Yeah, that's where firefly comes from, doesn't it.
I've got the base station plugged in, and the handset connected to
it,
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