Re: Re: [asterisk-users] Asterisk accepting calls to fast
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/g5/56830990,10,r) exten => 5683091,3,Hangup to exten => 5683091,2,Set(TIMEOUT(digit)=10) exten => 5683091,2,Answer() exten => 5683091,3,DIAL(ZAP/g5/56830990,10,r) exten => 5683091,4,Hangup() Asterisk has already chosen the wrong extension. The server has to wait for the complete number before it starts to look for any extension.I guess I need some kind of global timeout value. Stefan > > 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 => 56830910,4,hangup > > exten => 56830910,103,VoiceMail,b20 > > exten => 56830910,104,hangup > > > > exten => 5683091,1,Answer() > > exten => 5683091,2,DIAL(ZAP/g5/56830990,10,r) > > exten => 5683091,3,Hangup > > > > The problem now is, that sometimes (maybe when the caller doesn't hit the > > buttons fast enough) asterisk takes the extension for 5683091, although > > the 0 is still coming a little bit later. I'm not quite sure whether the > > delay in transferring the numbers is caused by the caller or by the > > telco. > > > > But is their a chance to tell asterisk to wait a little bit longer, > > before it starts searching the extensions.conf? Or do I have to tell the > > ISDN card to wait for the complete number, before it is forwarded to > > asterisk? > > -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk accepting calls to fast
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 => 56830910,4,hangup exten => 56830910,103,VoiceMail,b20 exten => 56830910,104,hangup exten => 5683091,1,Answer() exten => 5683091,2,DIAL(ZAP/g5/56830990,10,r) exten => 5683091,3,Hangup The problem now is, that sometimes (maybe when the caller doesn't hit the buttons fast enough) asterisk takes the extension for 5683091, although the 0 is still coming a little bit later. I'm not quite sure whether the delay in transferring the numbers is caused by the caller or by the telco. But is their a chance to tell asterisk to wait a little bit longer, before it starts searching the extensions.conf? Or do I have to tell the ISDN card to wait for the complete number, before it is forwarded to asterisk? Software & hardware: SuSE 10.0 Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1p chan_capi-0.7.0 divas4linux_EICON-106.20-1 Eicon Networks Corporation Diva Server 4BRI Rev 2 Thanks for your help & hints, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to transfer a caller out of a queue ?
Hi Rick, Am Montag, 18. September 2006 21:30 schrieb Rick Smith: > can't the agent just transfer the caller to another extension, whether that > be another queue or a person ? > yes, that's the easy part. But my client wants the caller (!) to be able to transfer himself into another 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 > Stefan-Michael. Guenther (in-put GbR) Sent: Monday, September 18, 2006 3:19 > PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] how to transfer a caller out of a queue ? > > 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 in the context where the call came in. In my configuration this > means, that my office phone is ringing: > > exten => 1,1,Answer() > exten => 1,2,DIAL(CAPI/@8304499:8304498,30,tTr) > exten => 1,3,Hangup > > But in this case not the caller, but the agent has been transferred! > Isn't there a chance for the caller to stop the conversation e.g. because > the agent told him that he has called the wrong queue and that he should > dial #1 to get to the right queue or directly to another person? > > If the agent does this, the caller get's transfered to the office phone, as > expected. > > As far as I understand the documentation, the context that is assigned to a > queue in queue.conf is only valid before an agent has accepted the call. > > I'm still running Asterisk 1.0.6, which is the current version for SuSE > 9.3. Maybe Asterisk 1.2.x would help? > > Thanks for help & hints, > > Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to transfer a caller out of a queue ?
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 in the context where the call came in. In my configuration this means, that my office phone is ringing: exten => 1,1,Answer() exten => 1,2,DIAL(CAPI/@8304499:8304498,30,tTr) exten => 1,3,Hangup But in this case not the caller, but the agent has been transferred! Isn't there a chance for the caller to stop the conversation e.g. because the agent told him that he has called the wrong queue and that he should dial #1 to get to the right queue or directly to another person? If the agent does this, the caller get's transfered to the office phone, as expected. As far as I understand the documentation, the context that is assigned to a queue in queue.conf is only valid before an agent has accepted the call. I'm still running Asterisk 1.0.6, which is the current version for SuSE 9.3. Maybe Asterisk 1.2.x would help? Thanks for help & hints, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to transfer a caller out of a queue ?
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 in the context where the call came in. In my configuration this means, that my office phone is ringing: exten => 1,1,Answer() exten => 1,2,DIAL(CAPI/@8304499:8304498,30,tTr) exten => 1,3,Hangup But in this case not the caller, but the agent has been transferred! Isn't there a chance for the caller to stop the conversation e.g. because the agent told him that he has called the wrong queue and that he should dial #1 to get to the right queue or directly to another person? If the agent does this, the caller get's transfered to the office phone, as expected. As far as I understand the documentation, the context that is assigned to a queue in queue.conf is only valid before an agent has accepted the call. I'm still running Asterisk 1.0.6, which is the current version for SuSE 9.3. Maybe Asterisk 1.2.x would help? Thanks for help & hints, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Missing number 2 in "advanced options" of VM
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 hint brought me a step further.
Now option 2 appears in the advanced options but the callback funtion seems to
be buggy (or my installation is buggy):
Even if I wait till the number has been completly read and then press 1 to
let asterisk call the number, only fragments of the number are forwarded to
the callback context. Sometimes even no digits are forwarded.
I have set callback=cback, and this context looks like this:
[cback]
exten => _X.,1,Answer()
exten => _X.,2,NoOp(${EXTEN})
exten => _X.,3,DIAL(CAPI/g1/83086921:${EXTEN},30,r)
And here's the output form the cli after I pressed 1 to call back:
Playing 'vm-num-i-have' (language 'de')
-- Playing 'digits/0' (language 'de')
-- Playing 'digits/7' (language 'de')
-- Playing 'digits/2' (language 'de')
-- Playing 'digits/1' (language 'de')
-- Playing 'digits/8' (language 'de')
-- Playing 'digits/3' (language 'de')
-- Playing 'digits/0' (language 'de')
-- Playing 'digits/4' (language 'de')
-- Playing 'digits/4' (language 'de')
-- Playing 'digits/9' (language 'de')
-- Playing 'digits/8' (language 'de')
-- Playing 'vm-tocallnum' (language 'de')
-- Destination number is CID number '4498'
-- Placing outgoing call to extension '4498' in context 'cback' from
context 'local'
-- Playing 'vm-dialout' (language 'de')
Aug 28 20:26:27 WARNING[14794]: pbx.c:2357 __ast_pbx_run: Channel
'SIP/111-081a1cc0' sent into invalid extension '4498' in context 'cback', but
no invalid handler
Why does Asterisk strip all digits except 4498 and why doesn't _X. match
4498??
Thanks for your help,
Stefan
--
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
Schulungen Installationen
Beratung Support
Voice over IP - Lösungen
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Missing number 2 in "advanced options" of VM
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 misunderstand the posting? Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk load testing
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 internet bandwidth does a typical call requires? I heard >around 20Kbps with typical codecs, is that right? > we have been responsible for an Asterisk server ( Celeron 2 GHz, 256 MB) that was "treated" with the ABACUS 5000. More info on the ABACUS: http://www.spirentcom.com/analysis/technology.cfm?az-c=pl&media=7&ws=325&ss=111 The ABACUS simulated up to 1100 SIP clients with 550 SIP calls between these clients. I'm still waiting for a more detailed report from the consultant who operated the ABACUS. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI doesn't execute PHP5 script [SOLVED]
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 executed! I guess I have to compare the parameters in the
two configuration files to get the important detail.
>> AGI Tx >> agi_accountcode:
>> AGI Tx >> LI>
>> -- AGI Script test.php completed, returning 0
>> -- Executing Hangup("OSS/dsp", "") in new stack
>> << Hangup on console >>
>>
>> And it doesn't make a difference whether I use the dial command or a sip
>>phone
>> to call extension 111 in context [guenther]. Strange, isn't it?
>Yeah. That response is usually when things are not happening properly.
>
Matt, the ouput hasn't changed, although ist script is executed properly. Why
do you thing that this output shows a failure?
>Also are you using php -q?
>
No, this isn't required for php5 (taken from README.CLI):
* CLI is started up in quiet mode by default.
(-q switch kept for compatibility)
Thanks for your help and suggestions,
Stefan
--
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
Schulungen Installationen
Beratung Support
Voice over IP - Lösungen
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI doesn't execute PHP5 script
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 execute the script manually on the command line, it is executes
> >>> without an error, the file is there and the email, too.
> >
> > ^^^
> >
> >> Try running it from the command line and see what happens
> >
> > I guess you meant the test.php script, right?
> > Executing php5 scripts on the command line isn't a problem at all, only
> > when they are started through AGI.
>
> This particular script also?
>
> Are you using AGI DEBUG in console?
>
yes, I can execute test.php on the command line and it runs as expected.
Wenn I call it via AGI nothing happens.
Yes, the first mail contained the output of the script with "agi debug", "set
verbose 10", "set debug 10" set before. Here it is again:
asterisk*CLI> dial [EMAIL PROTECTED]
-- Executing Answer("OSS/dsp", "") in new stack
<< Console call has been answered >>
-- Executing AGI("OSS/dsp", "test.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
AGI Tx >> agi_request: test.php
AGI Tx >> agi_channel: OSS/dsp
AGI Tx >> agi_language: en
AGI Tx >> agi_type: Console
AGI Tx >> agi_uniqueid: asterisk-6958-1155024459.47
AGI Tx >> agi_callerid: unknown
AGI Tx >> agi_calleridname: unknown
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: unknown
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: guenther
AGI Tx >> agi_extension: 111
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >> LI>
-- AGI Script test.php completed, returning 0
-- Executing Hangup("OSS/dsp", "") in new stack
<< Hangup on console >>
And it doesn't make a difference whether I use the dial command or a sip phone
to call extension 111 in context [guenther]. Strange, isn't it?
Stefan
--
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
Schulungen Installationen
Beratung Support
Voice over IP - Lösungen
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI doesn't execute PHP5 script
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 > > 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 > > 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 execute the script manually on the command line, it is executes > > without an error, the file is there and the email, too. ^^^ > > > > Try running it from the command line and see what happens > I guess you meant the test.php script, right? Executing php5 scripts on the command line isn't a problem at all, only when they are started through AGI. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI doesn't execute PHP5 script
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
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 execute the script manually on the command line, it is execute without
an error, the file is there and the email, too.
##
;extensions.conf
;
[guenther]
exten => 111,1,Answer()
exten => 111,2,AGI(test.php)
exten => 111,3,Hangup
##
ls -l /var/lib/asterisk/agi-bin/test.php
-rwxr-xr-x 1 asterisk root 340 Aug 8 10:07 test.php
##
cat /var/lib/asterisk/agi-bin/test.php
#! /usr/bin/php5
\nX-OTRS-Queue:Misc\n";
$subject="Anruf vom SysAdmin";
mail("[EMAIL PROTECTED]",$subject,"Testemail","From: Asterisk
<[EMAIL PROTECTED]>");
?>
##
asterisk*CLI> dial [EMAIL PROTECTED]
-- Executing Answer("OSS/dsp", "") in new stack
<< Console call has been answered >>
-- Executing AGI("OSS/dsp", "test.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
AGI Tx >> agi_request: test.php
AGI Tx >> agi_channel: OSS/dsp
AGI Tx >> agi_language: en
AGI Tx >> agi_type: Console
AGI Tx >> agi_uniqueid: asterisk-6958-1155024459.47
AGI Tx >> agi_callerid: unknown
AGI Tx >> agi_calleridname: unknown
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: unknown
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: guenther
AGI Tx >> agi_extension: 111
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >> LI>
-- AGI Script test.php completed, returning 0
-- Executing Hangup("OSS/dsp", "") in new stack
<< Hangup on console >>
##
When I use a shell script instead of the PHP script, everything works as
expected, the file /tmp/asterisk is created:
cat /var/lib/asterisk/agi-bin/test.sh
#! /bin/bash
echo "1" > /tmp/asterisk
##
Obviously, PHP5 is the problem, but what's wrong with it?
Thanks for your help,
Stefan
--
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
Schulungen Installationen
Beratung Support
Voice-over-IP-Loesungen
--
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
Schulungen Installationen
Beratung Support
Voice over IP - Lösungen
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax tone detected, but no fax extension for CAPI
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 in the configuration menu of the EICON card: Is this error caused by the port where the fax comes in (ISDN4/port 4) or by the outgoing port (ISDN3/port 3)? It would be strange if it is port 3, because normal calls get in and out on this port without any problems. Thanks, Stefan P.S: I have spent about two weeks now on this topic, together with the really helpful support from EICON. If I finish this configuration successfully, I promise I will write a HOWTO on it and post the link on the list;-)) -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax tone detected, but no fax extension for CAPI
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.
>
chan_capi is version 0.6.5 and asterisk is version 1.2.9.1.
Here's a complete log with verbosity and debugging set to 4.
Should I worry about the three messages
chan_capi.c:3951 capi_signal_progress: wrong channel state to signal PROGRESS
and
ISDN4: too much voice to send for NCCI=0x61a04
and
Protocol error layer 1 (broken line or B-channel removed by signalling
protocol)
-- CONNECT_IND
(PLCI=0x1a04,DID=07218304493,CID=5683099,CIP=0x4,CONTROLLER=0x4)
== ISDN4: Incoming call '5683099' -> '07218304493'
-- ISDN4: CAPI/ISDN4/07218304493-2: 07218304493 matches in context faxout
== Started pbx on channel CAPI/ISDN4/07218304493-2
-- ISDN4: info element CALLED PARTY NUMBER
-- ISDN4: info element DSP
-- ISDN4: info element CHANNEL IDENTIFICATION 81
-- ISDN4: info element SETUP
== ISDN4: pbx already started on channel CAPI/ISDN4/07218304493-2
> CAPI devicestate requested for ISDN4/07218304493
-- Executing Answer("CAPI/ISDN4/07218304493-2", "") in new stack
== ISDN4: Answering for 07218304493
-- Executing Dial("CAPI/ISDN4/07218304493-2", "CAPI/g1/07218304493|10|r")
in new stack
> data = g1/07218304493
> parsed dialstring: 'g1' 'NULL' '07218304493' ''
> capi request group = 2
> parsed dialstring: 'g1' 'NULL' '07218304493' ''
== ISDN3: Call CAPI/ISDN3/07218304493-3 (pres=0x00, ton=0x00)
-- Called g1/07218304493
== ISDN4: Requested RINGING-Indication for CAPI/ISDN4/07218304493-2
Aug 6 12:41:33 WARNING[21247]: chan_capi.c:3951 capi_signal_progress: wrong
channel state to signal PROGRESS
-- ISDN4: attempting ALERT in state 6
> CAPI devicestate requested for ISDN4/07218304493
> CAPI devicestate requested for ISDN3/07218304493
> CAPI devicestate requested for ISDN3/07218304493
== ISDN4: Setting up DTMF detector (PLCI=0x1a04, flag=1)
> ISDN4: DTMF conf(PLCI=0x1a04)
> ISDN4: too much voice to send for NCCI=0x61a04
-- ISDN3: received CONNECT_CONF PLCI = 0x303
-- ISDN3: info element CHANNEL IDENTIFICATION 89
-- ISDN3: info element CALL PROCEEDING
-- CAPI/ISDN3/07218304493-3 is proceeding passing it to
CAPI/ISDN4/07218304493-2
-- ISDN3: info element Date/Time 06/08/06 12:42
-- ISDN3: info element CONNECT
== ISDN3: Setting up DTMF detector (PLCI=0x303, flag=1)
-- CAPI/ISDN3/07218304493-3 answered CAPI/ISDN4/07218304493-2
== ISDN4: Requested Indication-STOP for CAPI/ISDN4/07218304493-2
-- Attempting native bridge of CAPI/ISDN4/07218304493-2 and
CAPI/ISDN3/07218304493-3
== ISDN4:ISDN3 Requested native bridge for CAPI/ISDN4/07218304493-2 and
CAPI/ISDN3/07218304493-3
> CAPI devicestate requested for ISDN3/07218304493
> ISDN3: DTMF conf(PLCI=0x303)
> ISDN4: too much voice to send for NCCI=0x61a04
-- ISDN3: info element FACILITY
-- ISDN3: info element CHARGE in UNITS
-- ISDN3: info element FACILITY
> ISDN3: c_dtmf = Yg
-- Fax tone detected, but no fax extension for CAPI/ISDN3/07218304493-3
-- ISDN3: info element DSP
-- ISDN3: info element INFORMATION
> CAPI INFO 0x3301: Protocol error layer 1 (broken line or B-channel
removed by signalling protocol)
== ISDN3: CAPI Hangingup
-- ISDN3: activehangingup (cause=0)
== ISDN4: CAPI Hangingup
== ISDN4: Interface cleanup PLCI=0x1a04
> CAPI devicestate requested for ISDN3/07218304493
-- ISDN3: info element FACILITY
-- ISDN3: info element DSP
-- ISDN3: info element CHARGE in UNITS
-- ISDN3: info element RELEASE
> CAPI INFO 0x3490: Normal call clearing
== ISDN3: Interface cleanup PLCI=0x303
I have set "faxdetect=off" in the [general] section of capi.conf and the
[faxout]-context looks like this (as recommended by Avi):
[faxout]
exten => _X.,1,Answer
exten => _X.,2,DIAL(CAPI/g1/${EXTEN},10,r)
exten => _X.,3,Congestion
exten => fax,1,Dial(CAPI/g1/${EXTEN},10,r)
Thanks for your help,
Stefan
--
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
Schulungen Installationen
Beratung Support
Voice over IP - Lösungen
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax tone detected, but no fax extension for CAPI
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 terminated before the fax has been sent with the
error message "Fax tone detected, but no fax extension for CAPI/ISDN3".
Google doesn't find a lot for this message. Does anyone of you have an idea
what's going wrong or what might be missing?
I have tested it with two different fax numbers and ISDN3 is able to answer
and start calls.
CONNECT_IND (PLCI=0x1a04,DID=07218304493,CID=5683099,CIP=0x4,CONTROLLER=0x4)
> ISDN4: msn='*' DNID='07218304493' MSN
== ISDN4: Incoming call '5683099' -> '07218304493'
-- ISDN4: CAPI/ISDN4/07218304493-0: 07218304493 matches in context faxout
== Started pbx on channel CAPI/ISDN4/07218304493-0
-- ISDN4: info element SETUP
== ISDN4: pbx already started on channel CAPI/ISDN4/07218304493-0
> CAPI devicestate requested for ISDN4/07218304493
-- Executing Answer("CAPI/ISDN4/07218304493-0", "") in new stack
== ISDN4: Answering for 07218304493
> CAPI devicestate requested for ISDN4/07218304493
-- Executing Dial("CAPI/ISDN4/07218304493-0", "CAPI/g1/07218304493|10|r")
in new stack
> data = g1/07218304493
> parsed dialstring: 'g1' 'NULL' '07218304493' ''
> capi request group = 2
> parsed dialstring: 'g1' 'NULL' '07218304493' ''
== ISDN3: Call CAPI/ISDN3/07218304493-1 (pres=0x00, ton=0x00)
-- Called g1/07218304493
== ISDN4: Requested RINGING-Indication for CAPI/ISDN4/07218304493-0
Aug 5 14:20:03 WARNING[9873]: chan_capi.c:3951 capi_signal_progress: wrong
channel state to signal PROGRESS
-- ISDN4: attempting ALERT in state 6
-- ISDN3: info element CALL PROCEEDING
-- CAPI/ISDN3/07218304493-1 is proceeding passing it to
CAPI/ISDN4/07218304493-0
-- CAPI/ISDN3/07218304493-1 answered CAPI/ISDN4/07218304493-0
== ISDN4: Requested Indication-STOP for CAPI/ISDN4/07218304493-0
-- Attempting native bridge of CAPI/ISDN4/07218304493-0 and
CAPI/ISDN3/07218304493-1
== ISDN4:ISDN3 Requested native bridge for CAPI/ISDN4/07218304493-0 and
CAPI/ISDN3/07218304493-1
> CAPI devicestate requested for ISDN3/07218304493
> ISDN3: c_dtmf = Y
-- Fax tone detected, but no fax extension for CAPI/ISDN3/07218304493-1
; capi.conf
;
[ISDN3]
incomingmsn=*
isdnmode=MSN
immediate=yes
context=isdnin
ntmode=no
controller=3
group=1 ;dialout group
callgroup=1
accountcode=ISDN3
devices=2
[ISDN4]
incomingmsn=*
immediate=yes
ntmode=yes
controller=4
group=9
callgroup=1
context=faxout
accountcode=ISDN4
devices=2
; extensions.conf
;
[faxout]
exten => _X.,1,Answer
exten => _X.,2,DIAL(CAPI/g1/${EXTEN},10,r)
exten => _X.,3,Congestion
Thanks for any help,
Stefan
--
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
Schulungen Installationen
Beratung Support
Voice-over-IP-Loesungen
--
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
Schulungen Installationen
Beratung Support
Voice over IP - Lösungen
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] file.c: Unexpected control subclass '14'
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 15:22:00 WARNING[31853] file.c: Unexpected control subclass '14' Jul 3 15:24:04 WARNING[31883] file.c: Unexpected control subclass '14' Jul 3 15:24:50 WARNING[31897] file.c: Unexpected control subclass '14' Asterisk 1.2.6 Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 or more ISDN cards: which comes first ??
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 is the first, so that I can setup capi.conf > > with the > > right entries? > > > > Thanks for your help, > > lspci should tell you... > That's easy, here's the output: 00:09.0 Network controller: Eicon Networks Corporation Diva Server 2FX (rev 01) 00:0a.0 Network controller: Eicon Networks Corporation Diva Server 4BRI Rev 2 (rev 01) And now let's get to the interesting and more difficult part: According to the output of lspci, the capi.conf should look like as follows, right? [isdn1] ; EICON 2 FX controller=1 group=1 devices=2 [isdn2] ; EICON 4 BRI 1st port group=2 controller=2 devices=2 [isdn3] ; EICON 4 BRI 2nd port group=2 controller=3 devices=2 [isdn4] ; EICON 4 BRI 3rd port group=2 controller=4 devices=2 [isdn5] ; EICON 4 BRI 4th port group=2 controller=5 devices=2 The 2FX is not connected to an ISDN line, all ports of the 4 BRI are connected in TE mode. I guess all the other parameters of this file aren't important for my problem. The thing is, that [isdn2] is dead, but when I configure a section [isdn6], I can use this controller for outgoing and incoming calls. There isn't a mistake in the configuration of the 4 BRI card itself otherwise I couldn't use ISDN3-ISDN6. The question is, what happened to ISDN2? Thanks for help & hints. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM Softphone on windows 2000
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 normal user, all values are resettet after restarting the software. This only happens when we use Win2k not with XP. Thanks for any help or hint. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 or more ISDN cards: which comes first ??
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-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] Voicemail records nonsense, but record() works (??)
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@lists.digium.com > Sent: 6/10/06 8:47 AM > Subject: [Asterisk-Users] Voicemail records nonsense, but record() works > (??) > > 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 => 83086921,103,VoiceMail,b111 > exten => 83086921,104,Hangup > > /etc/asterisk/voicemail.conf > [default] > language=de > 111 => 111,Mailbox 111,[EMAIL PROTECTED] > > The mailbox starts, I hear the intro and speak my message. In the CLI I can > see that the message has been recorded and I get the recorded message via > mail. > > But when I listen to the recorded messages or call the mailbox, I either > hear nothing or just a short cracking sound. At least the length of the > message is correct. If have tried to record the message with gsm, wav or > wav49, the result is always the same. > > When I use the record() application to record a gsm file, everything is > okay. > > I obviously made something wrong when configuring the voicemail system. > > Can someone give me a hint what's going wrong? > > Thanks for your help, > > stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] Voicemail records nonsense, but record() works (??)
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@lists.digium.com > Sent: 6/10/06 8:47 AM > Subject: [Asterisk-Users] Voicemail records nonsense, but record() works > (??) > > 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 => 83086921,103,VoiceMail,b111 > exten => 83086921,104,Hangup > > /etc/asterisk/voicemail.conf > [default] > language=de > 111 => 111,Mailbox 111,[EMAIL PROTECTED] > > The mailbox starts, I hear the intro and speak my message. In the CLI I can > see that the message has been recorded and I get the recorded message via > mail. > > But when I listen to the recorded messages or call the mailbox, I either > hear nothing or just a short cracking sound. At least the length of the > message is correct. If have tried to record the message with gsm, wav or > wav49, the result is always the same. > > When I use the record() application to record a gsm file, everything is > okay. > > I obviously made something wrong when configuring the voicemail system. > > Can someone give me a hint what's going wrong? > > Thanks for your help, > > stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Bug in Voicemail ??
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, [EMAIL PROTECTED] > > Hope it helps. > Thanks, I will compare it to my configuration. > One question to you, > you say you call the malbox, how do you do that? which extension do i > have to call to a ccess mailboxes? > You can define the extension as you like, here's my configuration: exten => 11101,1,Ringing exten => 11101,2,Wait(2) exten => 11101,3,VoiceMailMain,s111 exten => 22201,1,Ringing exten => 22201,2,Wait(2) exten => 22201,3,VoiceMailMain 11101 redirects your call directly to the mailbox 111, without asking for a password. 22201 wil ask you for the number of the mailbox and the password. Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bug in Voicemail ??
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 => 83086921,103,VoiceMail,b111 exten => 83086921,104,Hangup /etc/asterisk/voicemail.conf [default] language=de 111 => 111,Mailbox 111,[EMAIL PROTECTED] The mailbox starts, I hear the intro and speak my message. In the CLI I can see that the message has been recorded and I get the recorded message via mail. But when I listen to the recorded messages or call the mailbox, I either hear nothing or just a short cracking sound. At least the length of the message is correct. If have tried to record the message with gsm, wav or wav49, the result is always the same. When I use the record() application to record a gsm file, everything is okay. I obviously made something wrong when configuring the voicemail system. Can someone give me a hint what's going wrong? Thanks for your help, stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail records nonsense, but record() works (??)
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 => 83086921,103,VoiceMail,b111 exten => 83086921,104,Hangup /etc/asterisk/voicemail.conf [default] language=de 111 => 111,Mailbox 111,[EMAIL PROTECTED] The mailbox starts, I hear the intro and speak my message. In the CLI I can see that the message has been recorded and I get the recorded message via mail. But when I listen to the recorded messages or call the mailbox, I either hear nothing or just a short cracking sound. At least the length of the message is correct. If have tried to record the message with gsm, wav or wav49, the result is always the same. When I use the record() application to record a gsm file, everything is okay. I obviously made something wrong when configuring the voicemail system. Can someone give me a hint what's going wrong? Thanks for your help, stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Benchmarking an Asterisk Server with 14k users
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=111&wt=2) so simulate 14k users, calling each others, in other words 7k simultaneous calls. His main aim is to compare Cisco and Siemens, but he asked me whether we would be interested in testing the Asterisk server, too. Of course, I agreed! And I asked him whether we could publish the results, e.g. on this mailing list - he agreed! Well, asterisk experts, now it's your turn. Can you give me hints/information on how to optimize the asterisk system? And are there special details that you want me to test, besides the fact how many calls the system can handle? The more input you give me, the more output I can give back. The test will take place at the end of april. To make it clear: We don't want to compare the three system against each other. The asterisk server is running on a completely different hardware. We just have the chance to simulate a large number of callers on an asterisk system, we don't have to pay for the test and we can keep/publish the results -great! ;-)) Waiting for suggestions, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk management interface
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 > , if there is already AMP installed > Thanks > Joseph John > I'm afraid, but I'm sure that this won't work. I only had a quick look at the AMP before I decided to use the PBX Manager, so I don't know how AMP deals with the configuration files. The PBX Manager installs a number of include files which are necessary for editing the configuration files via the gui. I don't know whether the AMP does something like this, too, but I doubt that you can use both at the same day. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk management interface
Hi Nick, the english documentation is included in the webmin module, you can access it, when you click on the "help" link in the upper left corner. If you prefer a german documentation: http://www.in-put.de/voice-over-ip/PBX-Manager_4-0_de.pdf Stefan Am Donnerstag, 2. März 2006 10:29 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 > > > > It's based on webmin and well documented. > > > > Stefan > > Hi Stefan. What documentation have you found for Thirdlane's PBX Manager? > -- Nick > e: [EMAIL PROTECTED] > p: +61 7 5591 3588 > f: +61 7 5591 6588 > > If you receive this email by mistake, please notify us and do not make any > use of the email. We do not waive any privilege, confidentiality or > copyright associated with it. -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk management interface
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 PROTECTED]> > Subject: [Asterisk-Users] asterisk management interface > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > Hi everyone, > i am face with an asterisk use management interface, at the pressent, i am > using AMP (asterisk Management Portal: > http://coalescentsystems.ca/index.php?option=com_content&task=view&id=31&It >emid=57 ). > Does anyone know a better and more documented management interface for * ? > Thanks -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
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 from EICON: http://www.eicon.de/de/products/MediaGateways/all-in-one.htm http://www.eicon.de/de/products/MediaGateways/v-series.htm Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web interface
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 webmin. That's why there is no button in the PBX Manager to choose the language. > I will rephrase the statement.. > > AMP hands down is STILL the best FREE asterisk manager... > ACK ;-)) Bye, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Web interface
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, 2006-01-30 at 01:29 +, Strain Jer wrote: > > I was searching thru the internet and I found a wide variety of different > > web interfaces for asterisks > > I was curious which one is best suited for asterisks. Thanks > > -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] which gui for asterisk on web
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/asterisk-pbx-manager.html It's a webmin module and available in english, spanish and german. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for Windows based Asterisk
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, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Subject: [Asterisk-Users] Eicon Diva Server query
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 for voice only (which > isn't a problem, going into an Asterisk box). Do these cards play nicely > with Linux (2.6 kernel) and Asterisk? Any tips/tricks/pitfalls? > EICON offers rpm and deb packages for all major distributions and versions. The installation and configuration of the drivers is fairly easy. Just add chan_capi_cm after the installation of the driver and you're done. We are using the 4BRI and the BRI cards and never had any problems. If you need any help or hints you may contact me off the list, too. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re :Re: [Asterisk-Users] Softphone with Lotus Notes support?
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 GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Softphone with Lotus Notes support?
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 EyePMedia (http://www.eyepmedia.com/) Thanks for any suggestions, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] How to configure Asterisk through webmin
> 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. Webmin is a tool to help > administer the rest of the server. > and Asterisk, too: Have a look at THIRD LANE ASTERISK PBX MANAGER http://www.thirdlane.com/opensource.htm#manager Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pattern for matching CALLERID
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', but no invalid handler Could it be, that pattern matching doesn't work here ? It want to route calls based on the prefix of the number, so if pattern matching doesn' t work, I would use SubString and GotoIf instead. Have a nice day, Stefan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: OT: How to reach Junghanns.net?
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) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 Klaus-Peter Junghanns <[EMAIL PROTECTED]> Good luck, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One phone ringing, one phone flashing ?
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 phone and the phone of his secretary. The secretary's phones should ring, but at his phone only a light should flash. ;-)) No, turning off the sound isn't the solution. This restriction should e.g. only apply, when it is an external call, internal calls should result in ringing both phones. I'm not quite sure, whether this could be a feature of asterisk or the phone or both together. Does anything of you successfully set up something like this or could recommend a phone that would help/support it? Thanks a lot in advance, Stefan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: Re: [Asterisk-Users] IAX or IAX2 ? [SOLVED]
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 understood, I have to enter the file name in > > modules.conf, right? But if I do this, I get the error > > > > chan_iax2.so]Oct 11 10:09:52 WARNING[2288]: loader.c:258 > > ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undefined > > symbol: ast_check_signature > > Oct 11 10:09:52 WARNING[2288]: loader.c:391 load_modules: Loading module > > chan_iax2.so failed! > > Are you sure that the file and the main asterisk binary are from the > same source (e.g: debian package)? > Yes, it was the same source and the line load => chan_iax2.so was right. But I had to add load => res_crypto.so before. I used grep -r ast_check_signature /usr/src/asterisk-1.0.9/* to find out which other module might use ast_check_signature and so I found res_crypto. Have a nice weekend, Stefan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] IAX or IAX2 ?
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-0.2.0-RC7k on Debian 3.1) via IAX. > > Does this connection work with IAX or IAX2? > > IAX2 > ok. > > 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 understood, I have to enter the file name in modules.conf, right? But if I do this, I get the error chan_iax2.so]Oct 11 10:09:52 WARNING[2288]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_check_signature Oct 11 10:09:52 WARNING[2288]: loader.c:391 load_modules: Loading module chan_iax2.so failed! during the start. Well, so Asterisk is using IAX2 without an entry in modules.conf ? Stefan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX or IAX2 ?
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 the error message chan_iax2.so]Oct 11 10:09:52 WARNING[2288]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_check_signature Oct 11 10:09:52 WARNING[2288]: loader.c:391 load_modules: Loading module chan_iax2.so failed! [modules.conf] [modules] autoload=yes noload => pbx_gtkconsole.so noload => pbx_kdeconsole.so noload => app_intercom.so noload => chan_modem.so noload => chan_modem_aopen.so noload => chan_modem_i4l.so noload => chan_modem_bestdata.so noload => chan_alsa.so noload => chan_oss.so noload => rate_engine.so noload => chan_zap.so load => res_features.so load => res_musiconhold.so load => chan_capi.so load => chan_sip.so load => chan_iax2.so [global] chan_capi.so=yes Does Asterisk automatically use IAX, but IAX2 only when it is loaded via modules.conf? Google doesn't return a lot of entries for "undefined symbol: ast_check_signature" Thanks for any hint, Stefan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as a GSM-Gateway? Possible or not??
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 Asterisk Box. I found another posting that this feature is under development, does anyone know anything about it's status? And, final question, can anyone recommend a PC card for a GSM gateway? Thanks for any hint. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
