Hi,
Am Donnerstag, 7. Dezember 2006 19:31 schrieb Forrest Beck:
> Have a look at TIMEOUT(digit)
>
> http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout
>
I don't see how this function could help me.
If I change
exten => 5683091,1,Answer()
exten => 5683091,2,DIAL(ZAP/g5/5683099
Hi,
the german telco Colt Telekom has assigned the phone number block 56830-xxx to
one of our customers. In the diaplan we have setup extensions like the
following ones:
exten => 56830910,1,Answer()
exten => 56830910,2,Dial(SIP/bduerring,10,tr)
exten => 56830910,3,VoiceMail,u20
exten => 5683091
nother context. The reason for this is, that the
caller must decide whether he wants to be transfered from a free support line
to a support line for which he would have to pay.
Stefan
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> S
Hi,
I would like to give a caller the chance to leave a queue after an agent has
already accepted the call.
The caller enters the queue by dialing 333:
[from-sip]
exten => 300,1,Answer()
exten => 300,2,Queue(q1|tT)
When the caller presses # and e.g. 1, asterisk is looking for this extension
i
Hi,
I would like to give a caller the chance to leave a queue after an agent has
already accepted the call.
The caller enters the queue by dialing 333:
[from-sip]
exten => 300,1,Answer()
exten => 300,2,Queue(q1|tT)
When the caller presses # and e.g. 1, asterisk is looking for this extension
i
Hi,
> > a few weeks ago someone mentioned a menu point "2" in the advanced
> > options of the voicemail menu, which allows a call back to the caller who
> > left the message.
>
> Feature needs to be enabled in the voicemail.conf
>
> callback=context
>
> I've personally never used it.
>
well, this
Hi,
a few weeks ago someone mentioned a menu point "2" in the advanced options of
the voicemail menu, which allows a call back to the caller who left the
message.
I have two asterisk servers running but none has this second menu point.
Is this a feature which has to be enabled or did I misunder
Hi,
Am Dienstag, 15. August 2006 00:28 schrieb
[EMAIL PROTECTED]:
>Hi,
> did anyone try do load-testing on asterisk, for sip channel calls?
>I want to have a rough estimate about - how many calls, an asterisk server,
>running on say dual 240 opteron with 1 GB memory, can handle?
>Also how much in
Hi,
I have solved it (but don't understand yet, why it works)!!!
SuSE 10.1 uses different configuration files for the cli version and the cgi
version of PHP5.
When I modify the first line in the script from
#! /usr/bin/php5
to
#! /usr/bin/php5 -c /etc/php5/cli/
the php scripts gets execute
Hi,
Am Mittwoch, 9. August 2006 08:09 schrieb Matt Riddell (NZ):
> >>> The problem is, as you can see from the output in the CLI, that
> >>> Asterisk claims that it executes the script, but nothing happens. It
> >>> doesn't create the file /tmp/asterisk and it doesn't send an email.
> >>> When I e
Hi,
Am Dienstag, 8. August 2006 13:41 schrieb Matt Riddell (NZ):
> Stefan-Michael. Guenther (in-put GbR) wrote:
> > Hi,
> >
> > I'm trying to start a PHP5 script via the AGI Interface.
> > The asterisk version is Asterisk 1.2.5-BRIstuffed-0.3.0-PRE-1k and I
Hi,
I'm trying to start a PHP5 script via the AGI Interface.
The asterisk version is Asterisk 1.2.5-BRIstuffed-0.3.0-PRE-1k and I followed
the instructions on
http://www.voip-info.org/tiki-print.php?page=Asterisk+AGI+php
The problem is, as you can see from the output in the CLI, that Asterisk
Hi,
> > Protocol error layer 1 (broken line or B-channel removed by signalling
> > protocol)
>
> This is the cause of your problem! Your physical ISDN connection is
> broken. Maybe your cross/NT connection is not setup correct.
>
okay, then one last question before I start testing all the options
Hello,
> The log doesn't show anything about the call is terminated.
> Anyway, the message "Fax tone detected, but no fax extension for" is just a
> notice. If you don't have an extension "fax" in your context, nothing else
> is done. With newer chan_capi you can disable this with faxdetect=off.
>
Hello,
I have a fax server with an AVM Fritzcard that is connected to port number 4
of an EICON DIVA Server 4 BRI. As you can see from the following debug
messages, asterisk is accepting the incoming fax call on ISDN4 and forwards
it to port number 3 (ISDN3).
But at the end the call is terminat
Hi,
should I care about this error message in /var/log/asterisk/message?
Jul 3 15:18:42 WARNING[31782] file.c: Unexpected control subclass '14'
Jul 3 15:19:38 WARNING[31792] file.c: Unexpected control subclass '14'
Jul 3 15:21:08 WARNING[31837] file.c: Unexpected control subclass '14'
Jul 3 1
Hi,
Am Donnerstag, 29. Juni 2006 09:46 schrieb Francesco Peeters:
> On Thu, June 29, 2006 8:40, Stefan-Michael. Guenther (in-put GbR) said:
> > Hello,
> >
> > I have setup an asterisk server with two EICON cards, a 4BRI and 2FX.
> >
> > How do I know, which card
Hello,
I'm currently testing the SNOM softphone for one of our clients.
Is anyone on this list using this software on Windows 2000 as a normal user?
When we configure the softphone as an administrator and restart the software,
the configured values stay the same.
But when we configure it as a no
Hello,
I have setup an asterisk server with two EICON cards, a 4BRI and 2FX.
How do I know, which card is the first, so that I can setup capi.conf with the
right entries?
Thanks for your help,
Stefan
--
in-put GbR - Das Linux-Systemhaus
Stefan-Mic
Hello Strom,
thanks for your reply, but I guess your answer is missing (???).
Bye,
Stefan
Am Donnerstag, 15. Juni 2006 02:18 schrieb Strom Carlson:
> -Original Message-
> From: "Stefan-Michael. Guenther (in-put GbR)" <[EMAIL PROTECTED]>
> To: asterisk-users@l
Hello Strom,
thanks for your reply, but I guess your answer is missing (???).
Bye,
Stefan
Am Donnerstag, 15. Juni 2006 02:18 schrieb Strom Carlson:
> -Original Message-
> From: "Stefan-Michael. Guenther (in-put GbR)" <[EMAIL PROTECTED]>
> To: asterisk-users@l
Hello Victor,
> Hi,
> I'm still a newbie, but try to help you,
>
THX ;-))
> And voicemail.conf part is:
> [general]
> format=wav49
> maxmessage=180
> minmessage=2
> maxsilence=2
> silencethreshold=150
> maxlogins=3
> [EMAIL PROTECTED]
> skipms=3000
>
> [victor]
> victor => 1234, Victor Moreno, [
Hello,
I have setup an Asterisk 1.2.7.1 system, with a working voicemail box:
/etc/asterisk/extensions.conf
exten => 83086921,1,Answer
exten => 83086921,2,Dial(SIP/stefan,5,r)
exten => 83086921,3,VoiceMail,u111
exten => 83086921,4,Hangup
exten => 83086921,1
Hello,
I have setup an Asterisk 1.2.7.1 system, with a working voicemail box:
/etc/asterisk/extensions.conf
exten => 83086921,1,Answer
exten => 83086921,2,Dial(SIP/stefan,5,r)
exten => 83086921,3,VoiceMail,u111
exten => 83086921,4,Hangup
exten => 83086921,1
Hello,
one of our clients is currently testing three installations:
- Cisco Callmanager 5
- Siemens HiPath 8000
- Asterisk
To get an impression how these system behave under heavy load, he's going to
use an ABACUS 5000 system
(http://www.spirentcom.com/analysis/technology.cfm?media=7&WS=325&SS
Hi,
Am Donnerstag, 2. März 2006 10:32 schrieb John Joseph:
> thanks for this info, I have some doubts
> If I had already installed AMP , but I want to have
> PBX Manger installed , so that I can use both of them
> and compare each other
>will it cause problem if I install PBX manager
> ,
29 schrieb Nick Hoffman:
> On Thu March 2 2006 19:22, "Stefan-Michael. Guenther (in-put GbR)"
>
> <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > what about the Asterisk PBX Manager:
> >
> > http://www.thirdlane.com/opensource.htm#manager
> &
Hi,
what about the Asterisk PBX Manager:
http://www.thirdlane.com/opensource.htm#manager
It's based on webmin and well documented.
Stefan
Am Donnerstag, 2. März 2006 09:55 schrieb
[EMAIL PROTECTED]:
> Message: 9
> Date: Thu, 2 Mar 2006 08:59:23 +0100
> From: "Dumpolid Exeplish" <[EMAIL PROTEC
Hi,
> > I'm looking for an interface card for termination of Euro-ISDN2 (BRI)
> > lines.
> > That is ISDN lines from the telco into my Asterisk box.
> >
> > Any recommendations, good/bad expiriences ?
> >
> > At present I'm looking at cards from BeroNet and Junghanns.
> >
I prefer the ISDN cards f
Hello Zac,
> There is 1 problem.. I only took 1 semester of German 15 years ago.
> Looked all over the page for the English button, but I could not find one.
> I did wake up 10 minutes ago, so I could still be blind.
>
the language of the module is influenced by the language you choosed for
webm
Hi,
the Asterisk PBX Manager is STILL the best... though a few are catching up
quickly ;-)))
(Another absolutely subjective opinion)
http://www.in-put.de/voice-over-ip/asterisk-pbx-manager.html
Stefan
> AMP hands down is STILL the best... though a few are catching up quickly
>
> On Mon, 2006-0
Hi,
> Message: 12
> Date: Wed, 25 Jan 2006 05:34:29 -0500
> From: "Steve Totaro" <[EMAIL PROTECTED]>
> Subject: RE: [Asterisk-Users] which gui for asterisk on web
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
>
what about this one:
http://www.in-put.de/voice-over-ip/as
Hi,
>Does anyone know of a Asterisk Manager Interface client application that can
>run from a Windows XP machine to manage Asterisk installed on a Linux
>Machine.
>
if you consider the IE to be a client application, you could use the Asterisk
PBX Manager from Thirdlane (www.thirdlane.com).
Bye
Hi Avi,
>
> I've given up on crappy passive ISDN cards and am heading into the wild
> world of real, Active Super Dooper Server boards. I have a choice of two
> Eicon Diva Server cards:
>
> Eicon Diva Server 4BRI
> Eicon Diva Server V-4BRI
>
> The V-4BRI is actually cheaper, but I'm guessing its f
Hi,
>Do you think there would be any interest in a softphone that supports
>LDAP ?
>
why not? You can use ldap commands to connect to Domino and MS ADS, so a
softphone with ldap capabilities sounds like quite a good idea to me.
Stefan
--
in-put G
Hi,
has anyone of you heard of a softphone or client that support Lotus Notes?
I just want to click on the telephone number of an account and my hard- or
softphone should get the call.
Something similar to the outlook clients from Thirdlane
(http://www.thirdlane.com/opensource.htm#dialer)
or E
> On Thu, November 3, 2005 17:46, nr k said:
> > Hi all
> > I configured asterisk and webmin.i dont know how to
> > integrate webmin with asterisk and how to access
> > asterisk
> > through webmin.pls do the needful.
> >
> > regards
> > ramakrishnan.n
>
> Asterisk is not managed through webmin. We
Hi,
I would like to use pattern matching in what some call the ex-girlfriend rule:
[demo]
exten => 830449/_0721.,1,Answer()
exten => 830449/_0721.,2,Dial(SIP/stefan,20,tr)
When I dial 830449, asterisk tells me:
Channel 'CAPI/ISDN1/830449-3' sent into invalid extension 's' in context
'default',
Hi Peter,
> Does anybody know how I could make contact with them other than the
> published phone/email on their webpage?
>
I can offer you the following details of Mr. Junghanns himself:
CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +
Hi,
well, some clients have strange ideas and wishes (at least to my mind).
Yesterday I gave a presentation about asterisk to a CEO.
At the end he asked me whether asterisk is able to do the following:
When a call for the CEO comes in, the calling number should be shown on the
display of his ph
Hi,
> > > > When I try to load chan_iax2.so, I get the error message
> > >
> > > The channel name is iax. Yet it provides commands such that begin with
> > > "iax2" and listens on port 4569.
> >
> > ??? In /usr/lib/asterisk/modules the name of the file ist chan_iax2.so
> > and as far as I understo
Hello,
> From: Tzafrir Cohen <[EMAIL PROTECTED]>
> Subject: Re: [Asterisk-Users] IAX or IAX2 ?
>
> IAX1 is probably hardly used anywhere. Chances are that where people
> write IAX they mean IAX2
>
ok.
> > I have a working connection between two Asterisk-Servers (Asterisk
> > 1.0.7-BRIstuffed-0.2.
Hi,
I have read the wiki entries on IAX(2), but I'm afraid, it still have some
questions:
I have a working connection between two Asterisk-Servers (Asterisk
1.0.7-BRIstuffed-0.2.0-RC7k on Debian 3.1) via IAX.
Does this connection work with IAX or IAX2?
When I try to load chan_iax2.so, I get
Hi,
although I have spent a lot of time on searching the wiki and Google, I didn't
find an answer to the question whether it is possible to use Asterisk as a
GSM-Gateway.
The wiki mentions the Ateus VoiceBlue Box, but I don't want another box but
integrate the GSM gateway directly into the Ast
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