subject:"\[Asterisk\-Users\] chan

Re: [asterisk-users] chan_capi audio quality issue

2012-12-16 Thread Valer Nur

Check http://www.voip-info.org/wiki/view/Asterisk+RTCP





 From: Léopold Baillard 
To: asterisk-users@lists.digium.com 
Sent: Sunday, December 16, 2012 10:47 PM
Subject: Re: [asterisk-users] chan_capi audio quality issue
 
Le 16/12/2012 20:02, Valer Nur a écrit :
> From my experience, clicking noise can be originated from loss of audio 
> frames.
> 
> First, verify your CPU is not loaded,

It is not.

> then measure the frame loss to see if this is the source of the problem.
>

How can I do that?

> 
>
 
--

> *From:* Léopold Baillard 
> *To:* asterisk-users@lists.digium.com
> *Sent:* Wednesday, December 12, 2012 7:33 PM
> *Subject:* [asterisk-users] chan_capi audio quality issue
> 
> Hi everyone!
> 
> I'm installing in our small office a phone system using a Fritz!Card USB
> that I found in my attic. I'm using Asterisk 1.8.13, FreePBX 2.11,
> chan_capi 1.1.6.
> 
> Everything works great, I can place outgoing calls, ingoing calls work,
> internal calls too, voicemail, ...
> 
> The only problem is the audio quality when using chan_capi to place
> calls. The remote party can not hear me clearly and from the asterisk
> side, I hear a clicking noise (I can try to record it if that helps)
> repeating at close intervals but only when someone is speaking or when
> there's music (basically, when there is no silence).
> 
> I've skimmed Google for answers and came up with some old posts dating
> back to 2003/4/6/... speaking about the CAPI_MAX_B3_BLOCK_SIZE value in
> chan_capi.h. I tried to tweak that value and recompile several times.
> Each time, I have the same clicking noise but the interval changes. I
> can't get rid of the noise though.
> 
> This is currently preventing me from putting my setup in production and
> I'd really like to find a solution for this.
> 
> I'll be glad to hear any pointers you can throw at me. I can basically
> try everything as this is not in prod yet.
> 
> Thanks a lot in advance!
> 
> -- 
> Léopold Baillard
> - Administrateur de Léoserveur -
> http://www.leoserveur.org/
> 
> Courriel : leobaill...@leoserveur.org <mailto:leobaill...@leoserveur.org>
> Téléphone [FR] : + 33 (0) 6 20 32 16 32
> Téléphone [DE] : + 49 (0) 151 21 40 55 46
> GPG : 59C6 1CCA 2343 8DE4 D4FF D96A BC55 4A21 3B90 C658
> 
> 
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Courriel : leobaill...@leoserveur.org
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Re: [asterisk-users] chan_capi audio quality issue

2012-12-16 Thread Léopold Baillard

Le 16/12/2012 20:02, Valer Nur a écrit :
> From my experience, clicking noise can be originated from loss of audio 
> frames.
> 
> First, verify your CPU is not loaded,

It is not.

> then measure the frame loss to see if this is the source of the problem.
>

How can I do that?

> 
> --
> *From:* Léopold Baillard 
> *To:* asterisk-users@lists.digium.com
> *Sent:* Wednesday, December 12, 2012 7:33 PM
> *Subject:* [asterisk-users] chan_capi audio quality issue
> 
> Hi everyone!
> 
> I'm installing in our small office a phone system using a Fritz!Card USB
> that I found in my attic. I'm using Asterisk 1.8.13, FreePBX 2.11,
> chan_capi 1.1.6.
> 
> Everything works great, I can place outgoing calls, ingoing calls work,
> internal calls too, voicemail, ...
> 
> The only problem is the audio quality when using chan_capi to place
> calls. The remote party can not hear me clearly and from the asterisk
> side, I hear a clicking noise (I can try to record it if that helps)
> repeating at close intervals but only when someone is speaking or when
> there's music (basically, when there is no silence).
> 
> I've skimmed Google for answers and came up with some old posts dating
> back to 2003/4/6/... speaking about the CAPI_MAX_B3_BLOCK_SIZE value in
> chan_capi.h. I tried to tweak that value and recompile several times.
> Each time, I have the same clicking noise but the interval changes. I
> can't get rid of the noise though.
> 
> This is currently preventing me from putting my setup in production and
> I'd really like to find a solution for this.
> 
> I'll be glad to hear any pointers you can throw at me. I can basically
> try everything as this is not in prod yet.
> 
> Thanks a lot in advance!
> 
> -- 
> Léopold Baillard
> - Administrateur de Léoserveur -
> http://www.leoserveur.org/
> 
> Courriel : leobaill...@leoserveur.org <mailto:leobaill...@leoserveur.org>
> Téléphone [FR] : + 33 (0) 6 20 32 16 32
> Téléphone [DE] : + 49 (0) 151 21 40 55 46
> GPG : 59C6 1CCA 2343 8DE4 D4FF D96A BC55 4A21 3B90 C658
> 
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com 
> <http://www.api-digital.com/> --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
> 
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-- 
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- Administrateur de Léoserveur -
http://www.leoserveur.org/

Courriel : leobaill...@leoserveur.org
Téléphone [FR] : +33 (0) 6 20 32 16 32
Téléphone [DE] : +49 (0) 151 21 40 55 46
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Re: [asterisk-users] chan_capi audio quality issue

2012-12-16 Thread Valer Nur

>From my experience, clicking noise can be originated from loss of audio frames.

First, verify your CPU is not loaded, then measure the frame loss to see if 
this is the source of the problem.






 From: Léopold Baillard 
To: asterisk-users@lists.digium.com 
Sent: Wednesday, December 12, 2012 7:33 PM
Subject: [asterisk-users] chan_capi audio quality issue
 
Hi everyone!

I'm installing in our small office a phone system using a Fritz!Card USB
that I found in my attic. I'm using Asterisk 1.8.13, FreePBX 2.11,
chan_capi 1.1.6.

Everything works great, I can place outgoing calls, ingoing calls work,
internal calls too, voicemail, ...

The only problem is the audio quality when using chan_capi to place
calls. The remote party can not hear me clearly and from the asterisk
side, I hear a clicking noise (I can try to record it if that helps)
repeating at close intervals but only when someone is speaking or when
there's music (basically, when there is no silence).

I've skimmed Google for answers and came up with some old posts dating
back to 2003/4/6/... speaking about the CAPI_MAX_B3_BLOCK_SIZE value in
chan_capi.h. I tried to tweak that value and recompile several times.
Each time, I have the same clicking noise but the interval changes. I
can't get rid of the noise though.

This is currently preventing me from putting my setup in production and
I'd really like to find a solution for this.

I'll be glad to hear any pointers you can throw at me. I can basically
try everything as this is not in prod yet.

Thanks a lot in advance!

-- 
Léopold Baillard
- Administrateur de Léoserveur -
http://www.leoserveur.org/

Courriel : leobaill...@leoserveur.org
Téléphone [FR] : + 33 (0) 6 20 32 16 32
Téléphone [DE] : + 49 (0) 151 21 40 55 46
GPG : 59C6 1CCA 2343 8DE4 D4FF D96A BC55 4A21 3B90 C658


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[asterisk-users] chan_capi audio quality issue

2012-12-12 Thread Léopold Baillard

Hi everyone!

I'm installing in our small office a phone system using a Fritz!Card USB
that I found in my attic. I'm using Asterisk 1.8.13, FreePBX 2.11,
chan_capi 1.1.6.

Everything works great, I can place outgoing calls, ingoing calls work,
internal calls too, voicemail, ...

The only problem is the audio quality when using chan_capi to place
calls. The remote party can not hear me clearly and from the asterisk
side, I hear a clicking noise (I can try to record it if that helps)
repeating at close intervals but only when someone is speaking or when
there's music (basically, when there is no silence).

I've skimmed Google for answers and came up with some old posts dating
back to 2003/4/6/... speaking about the CAPI_MAX_B3_BLOCK_SIZE value in
chan_capi.h. I tried to tweak that value and recompile several times.
Each time, I have the same clicking noise but the interval changes. I
can't get rid of the noise though.

This is currently preventing me from putting my setup in production and
I'd really like to find a solution for this.

I'll be glad to hear any pointers you can throw at me. I can basically
try everything as this is not in prod yet.

Thanks a lot in advance!

-- 
Léopold Baillard
- Administrateur de Léoserveur -
http://www.leoserveur.org/

Courriel : leobaill...@leoserveur.org
Téléphone [FR] : + 33 (0) 6 20 32 16 32
Téléphone [DE] : + 49 (0) 151 21 40 55 46
GPG : 59C6 1CCA 2343 8DE4 D4FF D96A BC55 4A21 3B90 C658



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Re: [asterisk-users] chan_capi audio weirdness

2012-02-23 Thread Armin Schindler


Hello Arik,

On 02/18/2012 07:28 PM, Arik Raffael Funke wrote:

On 15/02/2012 22:53, Armin Schindler wrote:

I hear no progress indication. EVEN when using the r-option of the dial
command. It works however with
exten => _X.,1,Answer
exten => _X.,n,Dial(CAPI/contr1/12345)


in NT mode, the B-channel is not activated automatically. You have to signal
the TE side that early-B3 data is available. Then the TE side can activate
the B-channel. If the NT-side is chan_capi, use
exten => _X.,n,capicommand(progress)
without "Answer" before Dial().
Also, when using Dial() with chan_capi, you should use /b or /B option
in Dial() to get early-B3 from that other side too.
See README of chan_capi package for more details.


Thank you for your help. I did look at the chan_capi README, but I am afraid I
do not know enough about the capi protocol to make sense of everything.

I tried what you suggested - but without any luck. To make sure I did not
misunderstand you, I now have:
exten => _X.,1,capicommand(progress)
exten => _X.,n,Dial(CAPI/capi_intern/12345/b)


The capicommand(progress) is here not needed because the dialed line should
get the progress (because of /b) which is forwarded to the calling channel.
capicommand(progress) is needed e.g. when you want to play something before
that.


Used to call either from intern->intern or alternatively intern->extern. In
neither case do I get progress indication (i.e. ringing tones) as I did when
answering the channel before dialing.

Could it be that my hardware is simply behaving unexpectedly? After all, it's
not really a traditional capi card but an embedded device. Or am I still doing
something wrong?


Maybe. But I cannot tell without a capi debug log.

Armin

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Re: [asterisk-users] chan_capi audio weirdness

2012-02-19 Thread Arik Raffael Funke

Hi Armin,

On 18/02/2012 19:28, Arik Raffael Funke wrote:

in NT mode, the B-channel is not activated automatically. You have to
signal
the TE side that early-B3 data is available. Then the TE side can
activate
the B-channel. If the NT-side is chan_capi, use
exten => _X.,n,capicommand(progress)

>

I tried what you suggested - but without any luck. To make sure I did
not misunderstand you, I now have:
exten => _X.,1,capicommand(progress)
exten => _X.,n,Dial(CAPI/capi_intern/12345/b)

To help identification of the problem, my console prints the following 
after capicommand(progress):

[Feb 19 10:45:08] WARNING[3483]: chan_capi.c:4972 show_capi_conf_error: 
ISDN_INTERN#02: conf_error 0x2001 PLCI=0x103 
Command=SELECT_B_PROTOCOL_CONF,0x8495
   > ISDN_INTERN#02: CAPI INFO 0x2001: Message not supported in 
current state

Regards,
Arik

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Re: [asterisk-users] chan_capi audio weirdness

2012-02-18 Thread Arik Raffael Funke


Hello Armin,

On 15/02/2012 22:53, Armin Schindler wrote:

I hear no progress indication. EVEN when using the r-option of the dial
command. It works however with
 exten =>  _X.,1,Answer
 exten =>  _X.,n,Dial(CAPI/contr1/12345)


in NT mode, the B-channel is not activated automatically. You have to signal
the TE side that early-B3 data is available. Then the TE side can activate
the B-channel. If the NT-side is chan_capi, use
  exten =>  _X.,n,capicommand(progress)
without "Answer" before Dial().
Also, when using Dial() with chan_capi, you should use /b or /B option
in Dial() to get early-B3 from that other side too.
See README of chan_capi package for more details.


Thank you for your help. I did look at the chan_capi README, but I am 
afraid I do not know enough about the capi protocol to make sense of 
everything.


I tried what you suggested - but without any luck. To make sure I did 
not misunderstand you, I now have:

 exten =>  _X.,1,capicommand(progress)
 exten =>  _X.,n,Dial(CAPI/capi_intern/12345/b)

Used to call either from intern->intern or alternatively intern->extern. 
In neither case do I get progress indication (i.e. ringing tones) as I 
did when answering the channel before dialing.


Could it be that my hardware is simply behaving unexpectedly? After all, 
it's not really a traditional capi card but an embedded device. Or am I 
still doing something wrong?


Many thanks,
Arik


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Re: [asterisk-users] chan_capi audio weirdness

2012-02-15 Thread Armin Schindler

Hello Arik,

On 02/14/2012 12:49 PM, Arik Raffael Funke wrote:
> Hi,
> 
> I am trying to run asterisk on an AVM Fritz!Box Fon 7270 embedded DSL router.
> This works quite well after getting rid of the preinstalled phone server but I
> am encountering some unexpected behaviour.
> 
> Background: I am using two CAPI controllers provided by the hardware
> - one in MSN mode for dialling out and
> - one in NT-mode, (DID) for the internal S0-Bus
> 
> The problem is, I get no audio whatsoever until a channel is answered.
> Some of the symptoms of this are:
> - If I have an s-extension for the internal S0-Bus
> exten => s,1,Playtones(dial)
> I cannot hear the dialtone. It works however with:
> exten => s,1,Answer
> exten => s,n,Playtones(dial)
> 
> - Similarly if I dial from internal to external with the extension:
> exten => _X.,1,Dial(CAPI/contr1/12345)
> I hear no progress indication. EVEN when using the r-option of the dial
> command. It works however with
> exten => _X.,1,Answer
> exten => _X.,n,Dial(CAPI/contr1/12345)

in NT mode, the B-channel is not activated automatically. You have to signal
the TE side that early-B3 data is available. Then the TE side can activate
the B-channel. If the NT-side is chan_capi, use
 exten => _X.,n,capicommand(progress)
without "Answer" before Dial().
Also, when using Dial() with chan_capi, you should use /b or /B option
in Dial() to get early-B3 from that other side too.
See README of chan_capi package for more details.

Armin

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[asterisk-users] chan_capi audio weirdness

2012-02-14 Thread Arik Raffael Funke


Hi,

I am trying to run asterisk on an AVM Fritz!Box Fon 7270 embedded DSL 
router. This works quite well after getting rid of the preinstalled 
phone server but I am encountering some unexpected behaviour.


Background: I am using two CAPI controllers provided by the hardware
- one in MSN mode for dialling out and
- one in NT-mode, (DID) for the internal S0-Bus

The problem is, I get no audio whatsoever until a channel is answered.
Some of the symptoms of this are:
- If I have an s-extension for the internal S0-Bus
exten => s,1,Playtones(dial)
I cannot hear the dialtone. It works however with:
exten => s,1,Answer
exten => s,n,Playtones(dial)

- Similarly if I dial from internal to external with the extension:
exten => _X.,1,Dial(CAPI/contr1/12345)
I hear no progress indication. EVEN when using the r-option of the dial 
command. It works however with

exten => _X.,1,Answer
exten => _X.,n,Dial(CAPI/contr1/12345)


Has anybody seen this before?

Many thanks,
Arik


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Re: [asterisk-users] chan_capi install problems

2007-05-31 Thread Armin Schindler

On Thu, 31 May 2007, CSB wrote:
> > On Sat, 26 May 2007, CSB wrote:
> > > I have installed Asterisk 1.2.18 am am trying to install chan_capi.
> > > 
> > > The current RPM
> > > ftp://ftp.melware.net/Trixbox/chan_capi-1.0.1_1.2.14-1.i386.rpm
> > > installs but
> > 
> > This precompiled RPM is for the previous trixbox asterisk version 1.2.14.
> > A new RPM will follow soon...
> > 
> Do you have a rough idea of when?

I hope to have time on the weekend. There is already a bug-report for this:
 http://bugs.melware.net/mantis/view.php?id=34
I had to wait, because the trixbox system didn't provide the correct 
headers.

Armin

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Re: [asterisk-users] chan_capi install problems

2007-05-31 Thread CSB


On Sat, 26 May 2007, CSB wrote:

I have installed Asterisk 1.2.18 am am trying to install chan_capi.

The current RPM
ftp://ftp.melware.net/Trixbox/chan_capi-1.0.1_1.2.14-1.i386.rpm installs 
but


This precompiled RPM is for the previous trixbox asterisk version 1.2.14.
A new RPM will follow soon...


Do you have a rough idea of when?

Regards

Cameron 


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Re: [asterisk-users] chan_capi install problems

2007-05-26 Thread Armin Schindler

On Sun, 27 May 2007, CSB wrote:
> > > 
> > > The current RPM
> > > ftp://ftp.melware.net/Trixbox/chan_capi-1.0.1_1.2.14-1.i386.rpm
> > > installs but
> > 
> > This precompiled RPM is for the previous trixbox asterisk version 1.2.14.
> > A new RPM will follow soon...
> 
> I look forward to it.
> > 
> > If you want to compile chan-capi by yourself, you need to install all
> > dev-
> > packages to have the needed header files. I think this should do it:
> > yum -y install isdn4k-utils-devel asterisk-devel
> > 
> Having done that, I now get a message on asterisk startup:
> May 27 21:23:43 VERBOSE[4288] logger.c:  [chan_capi.so]May 27 21:23:43
> WARNING[4288] loader.c: /usr/lib/asterisk/modules/chan_capi.so: undefined
> symbol: ast_pickup_call
> May 27 21:23:43 WARNING[4288] loader.c: Loading module chan_capi.so failed!

new chan-capi uses ast_pickup_call too. But this function is provided by
module res_features. So you need to make sure to load res_features before 
chan-capi is loaded, e.g. in modules.conf:

[modules]
load=res_features.so
load=chan_capi.so


Armin
 
> > But if the trixbox asterisk version again has special patches applied
> > (something like jitterbuffer patch) which is not known to external
> > modules
> > like chan-capi, the compiled chan-capi may cause craches because it just
> > doesn't match with the configured asterisk header files.
> > 
> I am intending to use Trixbox but in the meantime for testing purposes have
> installed Asterisk from source.
> 
> Any further advice appreciated.
>
> Cameron 
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Re: [asterisk-users] chan_capi install problems

2007-05-26 Thread CSB



The current RPM
ftp://ftp.melware.net/Trixbox/chan_capi-1.0.1_1.2.14-1.i386.rpm installs 
but


This precompiled RPM is for the previous trixbox asterisk version 1.2.14.
A new RPM will follow soon...


I look forward to it.


If you want to compile chan-capi by yourself, you need to install all dev-
packages to have the needed header files. I think this should do it:
 yum -y install isdn4k-utils-devel asterisk-devel


Having done that, I now get a message on asterisk startup:
May 27 21:23:43 VERBOSE[4288] logger.c:  [chan_capi.so]May 27 21:23:43 
WARNING[4288] loader.c: /usr/lib/asterisk/modules/chan_capi.so: undefined 
symbol: ast_pickup_call

May 27 21:23:43 WARNING[4288] loader.c: Loading module chan_capi.so failed!


But if the trixbox asterisk version again has special patches applied
(something like jitterbuffer patch) which is not known to external modules
like chan-capi, the compiled chan-capi may cause craches because it just
doesn't match with the configured asterisk header files.

I am intending to use Trixbox but in the meantime for testing purposes have 
installed Asterisk from source.


Any further advice appreciated.

Cameron 


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Re: [asterisk-users] chan_capi install problems

2007-05-26 Thread Armin Schindler

On Sat, 26 May 2007, CSB wrote:
> I have installed Asterisk 1.2.18 am am trying to install chan_capi.
> 
> The current RPM
> ftp://ftp.melware.net/Trixbox/chan_capi-1.0.1_1.2.14-1.i386.rpm installs but

This precompiled RPM is for the previous trixbox asterisk version 1.2.14. 
A new RPM will follow soon...

> Asterisk dies on startup. The following appears in the log:
> May 27 03:28:18 asterisk1 kernel: divas: Diva Server V-4BRI-8 IRQ:7
> SerNo:25290
> May 27 03:28:18 asterisk1 kernel: divas: started with major 252
> May 27 03:54:17 asterisk1 init: Trying to re-exec init
> 
> The install notes say that the Asterisk version of the rpm must match so I
> guess that's the problem.

exactly.

> Downloading and making ftp://ftp.melware.net/chan-capi/chan_capi-1.0.1.tar.gz
> gives me a bunch of errors mostly "error: dereferencing pointer to incomplete
> type"

If you want to compile chan-capi by yourself, you need to install all dev- 
packages to have the needed header files. I think this should do it:
  yum -y install isdn4k-utils-devel asterisk-devel

But if the trixbox asterisk version again has special patches applied
(something like jitterbuffer patch) which is not known to external modules 
like chan-capi, the compiled chan-capi may cause craches because it just 
doesn't match with the configured asterisk header files.

Armin

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[asterisk-users] chan_capi install problems

2007-05-26 Thread CSB


I have installed Asterisk 1.2.18 am am trying to install chan_capi.

The current RPM 
ftp://ftp.melware.net/Trixbox/chan_capi-1.0.1_1.2.14-1.i386.rpm installs but 
Asterisk dies on startup. The following appears in the log:
May 27 03:28:18 asterisk1 kernel: divas: Diva Server V-4BRI-8 IRQ:7 
SerNo:25290

May 27 03:28:18 asterisk1 kernel: divas: started with major 252
May 27 03:54:17 asterisk1 init: Trying to re-exec init

The install notes say that the Asterisk version of the rpm must match so I 
guess that's the problem.


Downloading and making 
ftp://ftp.melware.net/chan-capi/chan_capi-1.0.1.tar.gz gives me a bunch of 
errors mostly "error: dereferencing pointer to incomplete type"


Any suggestions on what to do next are appreciated.

Regards

Cameron 


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Re: [asterisk-users] chan_capi and only one B channel usable?

2007-03-23 Thread Karsten Wemheuer

On 03/22/2007, Torge Szczepanek wrote:
> Hello list!
> 
> I have a Asterisk 1.2.10 running using the package from Backports.org
> for Debian Sarge.
> 
> I have setup chan_capi (0.6.5 from Backports) and it seems that I am
> only able to use on B-Channel.
> 
> When trying to place the second call I get:
> 
> CAPI INFO 0x34a2: No circuit / channel available
> 
> capi info shows:
> Contr1: 2 B channels total, 1 B channels free
> 
> And capi.conf is:
> 
> [ISDN1]
> msn=
> isdnmode=msn
> incomingmsn=*
> controller=1
> group=1
> softdtmf=on
> accountcode=
> context=remote
> echosquelch=1
> echocancel=yes
> echotail=64
> callgroup=1
> devices=2
> 
> Any ideas?

If You don't have any active calls from * using ISDN, there may be other
software using ISDN via the capi stack in Your box. The message
> capi info shows:
> Contr1: 2 B channels total, 1 B channels free
means, that there is one B-Channel used by CAPI (not neccessarily *).

HTH,
Karsten

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[asterisk-users] chan_capi and only one B channel usable?

2007-03-22 Thread Torge Szczepanek

Hello list!

I have a Asterisk 1.2.10 running using the package from Backports.org
for Debian Sarge.

I have setup chan_capi (0.6.5 from Backports) and it seems that I am
only able to use on B-Channel.

When trying to place the second call I get:

CAPI INFO 0x34a2: No circuit / channel available

capi info shows:
Contr1: 2 B channels total, 1 B channels free

And capi.conf is:

[ISDN1]
msn=
isdnmode=msn
incomingmsn=*
controller=1
group=1
softdtmf=on
accountcode=
context=remote
echosquelch=1
echocancel=yes
echotail=64
callgroup=1
devices=2

Any ideas?

Greetings Torge

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Re: [Asterisk-Users] chan_capi and bristuff

2006-10-27 Thread Armin Schindler

On Fri, 27 Oct 2006, Michiel van Baak wrote:
> On 23:11, Thu 26 Oct 06, Armin Schindler wrote:
> 
> > chan-capi-cm (chan-capi.org) is a complete rewritten version of chan-capi 
> > with more features and as far as I can tell, much more stable.
> > 
> > You do faxing with chan-capi 0.3.5? But this isn't faxing via CAPI, right?
> > As far as I know, chan-capi 0.3.5 does not support CAPI faxing.
> 
> You are correct. chan-capi 0.3.5 does not support faxing out
> of the box. But there's a patch out there that fixes this.
> It will add CapiAnswerFax(). This function saves a .sff
> (structured fax file) stream to disk which you can run
> through sfftobmp and stuff. We create PDF files with some
> commandline tools and it's rock stable (2 years without a
> missed/corrupted fax on a system that takes like 10 to 15
> faxes a week)
> Because it's working and we really believe in "if it aint
> broken dont fix it" we did not look into chan-capi-cm.

This feature provides chan-capi.org for a long time and since version 0.7
you can send faxes too.

> 
> My personal view on things: avoid PSTN/ISDN connections
> where possible and go with ITSP services. sangoma, quadbri,
> capi, tdm,... they all caused headaches where IAX simply
> works. Off course faxes wont work really good with ITSP but
> most of them have fax2email and email2fax.
> This is from a viewpoint of office PBX integration etc, not
> from ITSP viewpoint
> 

I don't agree here. I have a few servers running with chan-capi.org and
Eicon DIVA Server cards no problems at all.

Armin

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Re: [Asterisk-Users] chan_capi and bristuff

2006-10-26 Thread Michiel van Baak

On 23:11, Thu 26 Oct 06, Armin Schindler wrote:

> chan-capi-cm (chan-capi.org) is a complete rewritten version of chan-capi 
> with more features and as far as I can tell, much more stable.
> 
> You do faxing with chan-capi 0.3.5? But this isn't faxing via CAPI, right?
> As far as I know, chan-capi 0.3.5 does not support CAPI faxing.

You are correct. chan-capi 0.3.5 does not support faxing out
of the box. But there's a patch out there that fixes this.
It will add CapiAnswerFax(). This function saves a .sff
(structured fax file) stream to disk which you can run
through sfftobmp and stuff. We create PDF files with some
commandline tools and it's rock stable (2 years without a
missed/corrupted fax on a system that takes like 10 to 15
faxes a week)
Because it's working and we really believe in "if it aint
broken dont fix it" we did not look into chan-capi-cm.

My personal view on things: avoid PSTN/ISDN connections
where possible and go with ITSP services. sangoma, quadbri,
capi, tdm,... they all caused headaches where IAX simply
works. Off course faxes wont work really good with ITSP but
most of them have fax2email and email2fax.
This is from a viewpoint of office PBX integration etc, not
from ITSP viewpoint

-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

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Re: [Asterisk-Users] chan_capi and bristuff

2006-10-26 Thread Armin Schindler

On Thu, 26 Oct 2006, Michiel van Baak wrote:
> On 14:28, Thu 26 Oct 06, Tzafrir Cohen wrote:
> > On Thu, Oct 26, 2006 at 10:48:52AM +0200, Michiel van Baak wrote:
> > 
> > > the chan_capi is merged with BRIStuff. This means you no
> > > longer have to download and compile chan_capi manually when
> > > you want to use a CAPI board with bristuffed asterisk.
> > 
> > And while we're at it: How does the bristuff chan_capi compare the
> > chan_capi-cm?
> 
> I have no idea. I still use the patched chan_capi 0.3.5 and
> all I do with it is receiving faxes ;)
> 2 companies we installed the asterisk boxen for use
> chan_capi from bristuff 0.3.0 and they are happy with it (in
> combination with AVM Fritz!PCI cards using the avm driver)
> Since I have no experience with chan_capi-cm I cant tell you
> which one is better.

chan-capi-cm (chan-capi.org) is a complete rewritten version of chan-capi 
with more features and as far as I can tell, much more stable.

You do faxing with chan-capi 0.3.5? But this isn't faxing via CAPI, right?
As far as I know, chan-capi 0.3.5 does not support CAPI faxing.

Armin

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Re: [Asterisk-Users] chan_capi and bristuff

2006-10-26 Thread Michiel van Baak

On 14:28, Thu 26 Oct 06, Tzafrir Cohen wrote:
> On Thu, Oct 26, 2006 at 10:48:52AM +0200, Michiel van Baak wrote:
> 
> > the chan_capi is merged with BRIStuff. This means you no
> > longer have to download and compile chan_capi manually when
> > you want to use a CAPI board with bristuffed asterisk.
> 
> And while we're at it: How does the bristuff chan_capi compare the
> chan_capi-cm?

I have no idea. I still use the patched chan_capi 0.3.5 and
all I do with it is receiving faxes ;)
2 companies we installed the asterisk boxen for use
chan_capi from bristuff 0.3.0 and they are happy with it (in
combination with AVM Fritz!PCI cards using the avm driver)
Since I have no experience with chan_capi-cm I cant tell you
which one is better.
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

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Re: [Asterisk-Users] chan_capi and bristuff

2006-10-26 Thread Tzafrir Cohen

On Thu, Oct 26, 2006 at 10:48:52AM +0200, Michiel van Baak wrote:

> the chan_capi is merged with BRIStuff. This means you no
> longer have to download and compile chan_capi manually when
> you want to use a CAPI board with bristuffed asterisk.

And while we're at it: How does the bristuff chan_capi compare the
chan_capi-cm?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [Asterisk-Users] chan_capi and bristuff

2006-10-26 Thread Michiel van Baak

On 09:39, Thu 26 Oct 06, Olivier wrote:
> Hi,
> 
> Reading from www.voip-info.org, i can see that "Junghann's chan_capi is now
> part of bristuff  , as of
> version 0.3.0-pre".
> What does that really mean ?
> 
> Shall I understand I can share a Junghanns QuadBRI board between 2
> CAPI-enabled software (like a 0.3.0-pre bristuffed Asterisk for instance) ?
> If positive, what are the pros and cons of such solution ?
> 
> Except for legacy applications support (and this could be a very good
> reason), I can't figure out myself any benefit using CAPI for HFC cards for
> new setups as those cards don't embed any DSP hardware.

You mixed up 2 things.
The Junghanns QuadBRI board comes with a driver qozap.
This means it's a zaptel device. Not a CAPI device.

the chan_capi is merged with BRIStuff. This means you no
longer have to download and compile chan_capi manually when
you want to use a CAPI board with bristuffed asterisk.

As you can see those two things have nothing to do with
eachother. the quadbri is still a zap card. It's simply more
convenient when you want to support only one version of
asterisk and you have sites with quadbri and sites with CAPI
hardware. (or sites with both quadbri and capi hardware in
one machine)

cheers

-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

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[Asterisk-Users] chan_capi and bristuff

2006-10-26 Thread Olivier

Hi,Reading from www.voip-info.org, i can see that "Junghann's chan_capi is now part of bristuff , as of version 
0.3.0-pre".What does that really mean ?Shall I understand I can share a Junghanns QuadBRI board between 2 CAPI-enabled software (like a 0.3.0-pre bristuffed Asterisk for instance) ?If positive, what are the pros and cons of such solution ?
Except for legacy applications support (and this could be a very good reason), I can't figure out myself any benefit using CAPI for HFC cards for new setups as those cards don't embed any DSP hardware.Regards

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[Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem

2006-06-06 Thread Esteban Guana-Jarrin


Armin,

2. Incoming DTMF tones are not detected via this ISDN line and AVM Fritz 
card



Did you set for softdtmf/relaxdtmf?



Armin


I have tried with the following in capi.conf:

softdtmf=on  ;enable/disable software dtmf detection, recommended for 
AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf 
detection


This is the capi-in context in extensions.conf that i'm just using to test 
the digits detection


[capi-in]
exten => 99546476,1,Answer
exten => 99546476,2,DigitTimeout(10)
exten => 99546476,3,ResponseTimeout(5)
exten => 99546476,4,Read(Digits,enter-phone-number10,10)
exten => 99546476,5,NoOp(${Digits})
exten => t,1, Goto(99546476,4)
exten => t,2, Hangup

Below is the debug output and as you can see the variable Digits is empty 
(User entered ''). No digits were detected.


== ISDN1: Incoming call '' -> '99546476'
   -- Executing Answer("CAPI/ISDN1/99546476-18", "") in new stack
 == ISDN1: Answering for 99546476
   -- Executing DigitTimeout("CAPI/ISDN1/99546476-18", "10") in new stack
   -- Set Digit Timeout to 10
   -- Executing ResponseTimeout("CAPI/ISDN1/99546476-18", "5") in new stack
   -- Set Response Timeout to 5
   -- Executing Read("CAPI/ISDN1/99546476-18", 
"Digits|enter-phone-number10|10") in new stack

   -- Accepting a maximum of 10 digits.
   -- Playing 'enter-phone-number10' (language 'en')
   -- User entered ''
   -- Executing NoOp("CAPI/ISDN1/99546476-18", "") in new stack
   -- Timeout on CAPI/ISDN1/99546476-18
 == CDR updated on CAPI/ISDN1/99546476-18
   -- Executing Goto("CAPI/ISDN1/99546476-18", "99546476|4") in new stack
   -- Goto (capi-in,99546476,4)
   -- Executing Read("CAPI/ISDN1/99546476-18", 
"Digits|enter-phone-number10|10") in new stack

   -- Accepting a maximum of 10
   -- Playing 'enter-phone-number10' (language 'en')
   -- User entered ''
   -- Executing NoOp("CAPI/ISDN1/99546476-18", "") in new stack
   -- Timeout on CAPI/ISDN1/99546476-18

Any ideas?

Esteban

_
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Re: [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem

2006-06-06 Thread Armin Schindler

On Tue, 6 Jun 2006, Esteban Guana-Jarrin wrote:
> James and Armin,
> 
> > > Turn on asterisk debugging too. Capi seems to be working okay, maybe
> > > asterisk isn't picking up the call for some reason. Maybe:
> 
> > > asterisk -r
> > > set verbose 9
> > > set debug 9
> > > capi debug
> 
> > > then make an incoming call and copy the output into an email and send
> > > it
> > > to the list (unless it is really really long, then you may have to
> > > look
> > > for interesting bits).
> 
> > > u should see a message in there somewhere that tells you that either
> > > the capi driver is rejecting the call because it doesn't want to
> > > answer
> > > that msn (your earlier logs make that unlikelye), or that asterisk
> > > can't
> > > find an extension for it.
> 
> > > James
> 
> Thanks for your response I managed to get it working with the following as you
> suggested,
> 
> [capi-in]
> exten => 99546476,1,Dial(Sip/123,20)
> exten => 99546476,2,Voicemail(123)
> exten => 99546476,3,Hangup
> 
> Following is the debug output when it started working,
> 
> -- ISDN1: info element CHANNEL IDENTIFICATION 89
> INFO_IND ID=001 #0x09d5 LEN=0016
> Controller/PLCI/NCCI= 0x101
> InfoNumber  = 0xa1
> InfoElement = 
> 
> INFO_RESP ID=001 #0x09d5 LEN=0012
> Controller/PLCI/NCCI= 0x101
> 
> -- ISDN1: info element Sending Complete
> -- ISDN1: CAPI/ISDN1/99546476-16: 99546476 matches in context capi-in
> -- Executing Dial("CAPI/ISDN1/99546476-16", "Sip/123|20") in new stack
> == Started pbx on channel CAPI/ISDN1/99546476-16
> -- Called 123
> -- SIP/123-0b4d is ringing
> == ISDN1: Requested RINGING-Indication for CAPI/ISDN1/99546476-16

Looks good.
 
> I now have two other problems,
> 
> 1. Noise is quite loud using this line with asterisk

Maybe you should adapt the gains in capi.conf.

> 2. Incoming DTMF tones are not detected via this ISDN line and AVM Fritz card

Did you set for softdtmf/relaxdtmf?

Armin

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[Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem

2006-06-06 Thread Esteban Guana-Jarrin


James and Armin,


Turn on asterisk debugging too. Capi seems to be working okay, maybe
asterisk isn't picking up the call for some reason. Maybe:



asterisk -r
set verbose 9
set debug 9
capi debug



then make an incoming call and copy the output into an email and send it
to the list (unless it is really really long, then you may have to look
for interesting bits).



u should see a message in there somewhere that tells you that either
the capi driver is rejecting the call because it doesn't want to answer
that msn (your earlier logs make that unlikelye), or that asterisk can't
find an extension for it.



James


Thanks for your response I managed to get it working with the following as 
you suggested,


[capi-in]
exten => 99546476,1,Dial(Sip/123,20)
exten => 99546476,2,Voicemail(123)
exten => 99546476,3,Hangup

Following is the debug output when it started working,

-- ISDN1: info element CHANNEL IDENTIFICATION 89
INFO_IND ID=001 #0x09d5 LEN=0016
 Controller/PLCI/NCCI= 0x101
 InfoNumber  = 0xa1
 InfoElement = 

INFO_RESP ID=001 #0x09d5 LEN=0012
 Controller/PLCI/NCCI= 0x101

   -- ISDN1: info element Sending Complete
   -- ISDN1: CAPI/ISDN1/99546476-16: 99546476 matches in context capi-in
   -- Executing Dial("CAPI/ISDN1/99546476-16", "Sip/123|20") in new stack
 == Started pbx on channel CAPI/ISDN1/99546476-16
   -- Called 123
   -- SIP/123-0b4d is ringing
 == ISDN1: Requested RINGING-Indication for CAPI/ISDN1/99546476-16

I now have two other problems,

1. Noise is quite loud using this line with asterisk
2. Incoming DTMF tones are not detected via this ISDN line and AVM Fritz 
card


Any ideas on how to overcome these issues?

Esteban

_
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RE: [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem

2006-06-05 Thread James Harper

> My dial plan as shown below is,
> 
> [capi-in]
> exten => s,1,Dial(Sip/123,20)
> exten => s,2,Voicemail(123)
> exten => s,3,Hangup
> 
> I believe I should be able to receive calls with the above.

With immediate = yes then you should.

> I have also tried the following, and i get the same problem and debug
> output is the same.
> 
> [capi-in]
> exten => 99546476,1,Dial(Sip/123,20)
> exten => 99546476,2,Voicemail(123)
> exten => 99546476,3,Hangup
> 
> Any other ideas ???

Turn on asterisk debugging too. Capi seems to be working okay, maybe
asterisk isn't picking up the call for some reason. Maybe: 

asterisk -r
set verbose 9
set debug 9
capi debug

then make an incoming call and copy the output into an email and send it
to the list (unless it is really really long, then you may have to look
for interesting bits).

You should see a message in there somewhere that tells you that either
the capi driver is rejecting the call because it doesn't want to answer
that msn (your earlier logs make that unlikelye), or that asterisk can't
find an extension for it.

James
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Re: [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem

2006-06-05 Thread Armin Schindler

On Mon, 5 Jun 2006, Esteban Guana-Jarrin wrote:
> Thanks Armin
> 
> > > The call is rejected by Asterisk, so it looks like your dialplan
> > > has no rule for accepting calls to '99546476'.
> 
> > > Armin
> 
> 
> My dial plan as shown below is,
> 
> [capi-in]
> exten => s,1,Dial(Sip/123,20)
> exten => s,2,Voicemail(123)
> exten => s,3,Hangup
> 
> I believe I should be able to receive calls with the above.

No, 's' is not used. The called number must be used.
 
> I have also tried the following, and i get the same problem and debug output
> is the same.
> 
> [capi-in]
> exten => 99546476,1,Dial(Sip/123,20)
> exten => 99546476,2,Voicemail(123)
> exten => 99546476,3,Hangup
> 
> Any other ideas ???

That looks correct. Is the context= in capi.conf really set to capi-in ?

Armin

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[Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem

2006-06-05 Thread Esteban Guana-Jarrin


Thanks Armin


The call is rejected by Asterisk, so it looks like your dialplan
has no rule for accepting calls to '99546476'.



Armin



My dial plan as shown below is,

[capi-in]
exten => s,1,Dial(Sip/123,20)
exten => s,2,Voicemail(123)
exten => s,3,Hangup

I believe I should be able to receive calls with the above.

I have also tried the following, and i get the same problem and debug output 
is the same.


[capi-in]
exten => 99546476,1,Dial(Sip/123,20)
exten => 99546476,2,Voicemail(123)
exten => 99546476,3,Hangup

Any other ideas ???

Esteban

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Re: [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem

2006-06-05 Thread Armin Schindler

The call is rejected by Asterisk, so it looks like your dialplan
has no rule for accepting calls to '99546476'.

Armin


On Mon, 5 Jun 2006, Esteban Guana-Jarrin wrote:

> I have a problem receving calls via the ISDN line, using the followin
> components
> 
> Asterisk 1.0.9 with [EMAIL PROTECTED]
> chan_capi-cm-0.6
> AVM Fritz card
> datalink protocol = point to multimode
> 
> I can make calls out with no problems so the issue is only incoming calls.
> 
> When I make the call from an external line to the ISDN line connected to
> asterisk, I get a busy signal after about 5 seconds. I have read previous
> posts from the list and I made sure I have the settings in my capi.conf and
> extensions.conf according to all the suggestions and as shown below,
> 
> Capi.conf:
> 
> [general]
> nationalprefix=0
> internationalprefix=00
> rxgain=0.8
> txgain=0.8
> ulaw=yes;set this, if you live in u-law world instead of a-law
> 
> [ISDN1]  ;this example interface gets name 'ISDN1' and may be any
> ;name not starting with 'g' or 'contr'.
> 
> isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
>  ;when using NT-mode, ptp should be set in any case
> msn=0299546476
> incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * == any
> controller=1 ;capi controller number to use
> group=2  ;dialout group
> context=capi-in  ;context for incoming calls
> immediate=yes   ;immediate start of pbx with extension 's' if no digits were
> ;received on incoming call (no destination number yet)
> devices=2;number of concurrent calls on this controller
> ;(2 makes sense for single BRI, 30 for PRI)
> 
> In Extensions.conf
> 
> [capi-in]
> exten => s,1,Dial(Sip/123,20)
> exten => s,2,Voicemail(123)
> exten => s,3,Hangup
> 
> Also the Capi debug output with verbosity 15, is shown below. I can see that
> after the channel identification message and then the sending complete
> message, a DISCONNECT_IND comes straight after and it does not provide any
> reasons...
> 
> CAPI Debugging Enabled
> CONNECT_IND ID=001 #0x03a6 LEN=0037
> Controller/PLCI/NCCI= 0x101
> CIPValue= 0x10
> CalledPartyNumber   = 99546476
> CallingPartyNumber  = default
> CalledPartySubaddress   = default
> CallingPartySubaddress  = default
> BC  = <80 90 a3>
> LLC = default
> HLC = <91 81>
> AdditionalInfo  = default
> 
>-- CONNECT_IND (PLCI=0x101,DID=99546476,CID=(null),CIP=0x10,CONTROLLER=0x1)
> > ISDN1: msn='*' DNID='99546476' MSN
>  == ISDN1: Incoming call '' -> '99546476'
> INFO_IND ID=001 #0x03a7 LEN=0024
> Controller/PLCI/NCCI= 0x101
> InfoNumber  = 0x70
> InfoElement = 99546476
> 
> INFO_RESP ID=001 #0x03a7 LEN=0012
> Controller/PLCI/NCCI= 0x101
> 
>-- ISDN1: info element CALLED PARTY NUMBER
> > ISDN1: INFO_IND DID digits not used in this state.
> INFO_IND ID=001 #0x03a8 LEN=0016
> Controller/PLCI/NCCI= 0x101
> InfoNumber  = 0x18
> InfoElement = <89>
> 
> INFO_RESP ID=001 #0x03a8 LEN=0012
> Controller/PLCI/NCCI= 0x101
> 
>-- ISDN1: info element CHANNEL IDENTIFICATION 89
> INFO_IND ID=001 #0x03a9 LEN=0016
> Controller/PLCI/NCCI= 0x101
> InfoNumber  = 0xa1
> InfoElement = 
> 
> INFO_RESP ID=001 #0x03a9 LEN=0012
> Controller/PLCI/NCCI= 0x101
> 
>-- ISDN1: info element Sending Complete
> CONNECT_RESP ID=001 #0x03a9 LEN=0032
> Controller/PLCI/NCCI= 0x101
> Reject  = 0x1
> BProtocol
> B1protocol = 0x0
> B2protocol = 0x0
> B3protocol = 0x0
> B1configuration= default
> B2configuration= default
> B3configuration= default
> ConnectedNumber = default
> ConnectedSubaddress = default
> LLC = default
> AdditionalInfo
> BChannelinformation= default
> Keypadfacility = default
> Useruserdata   = default
> Facilitydataarray  = default
> 
> DISCONNECT_IND ID=001 #0x03aa LEN=0014
> Controller/PLCI/NCCI= 0x101
> Reason  = 0x0
> 
> DISCONNECT_RESP ID=001 #0x03aa LEN=0012
> Controller/PLCI/NCCI= 0x101
> 
> == ISDN1: CAPI Hangingup
> == ISDN1: Interface cleanup PLCI=0x101
> 
> I will apreciate your assistance
> 
> Esteban
> 
> _
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> 
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[Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem

2006-06-04 Thread Esteban Guana-Jarrin

I have a problem receving calls via the ISDN line, using the followin 
components


Asterisk 1.0.9 with [EMAIL PROTECTED]
chan_capi-cm-0.6
AVM Fritz card
datalink protocol = point to multimode

I can make calls out with no problems so the issue is only incoming calls.

When I make the call from an external line to the ISDN line connected to 
asterisk, I get a busy signal after about 5 seconds. I have read previous 
posts from the list and I made sure I have the settings in my capi.conf and 
extensions.conf according to all the suggestions and as shown below,


Capi.conf:

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
ulaw=yes;set this, if you live in u-law world instead of a-law

[ISDN1]  ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.

isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
 ;when using NT-mode, ptp should be set in any case
msn=0299546476
incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * == any
controller=1 ;capi controller number to use
group=2  ;dialout group
context=capi-in  ;context for incoming calls
immediate=yes   ;immediate start of pbx with extension 's' if no digits were
;received on incoming call (no destination number yet)
devices=2;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)

In Extensions.conf

[capi-in]
exten => s,1,Dial(Sip/123,20)
exten => s,2,Voicemail(123)
exten => s,3,Hangup

Also the Capi debug output with verbosity 15, is shown below. I can see that 
after the channel identification message and then the sending complete 
message, a DISCONNECT_IND comes straight after and it does not provide any 
reasons...


CAPI Debugging Enabled
CONNECT_IND ID=001 #0x03a6 LEN=0037
 Controller/PLCI/NCCI= 0x101
 CIPValue= 0x10
 CalledPartyNumber   = 99546476
 CallingPartyNumber  = default
 CalledPartySubaddress   = default
 CallingPartySubaddress  = default
 BC  = <80 90 a3>
 LLC = default
 HLC = <91 81>
 AdditionalInfo  = default

   -- CONNECT_IND 
(PLCI=0x101,DID=99546476,CID=(null),CIP=0x10,CONTROLLER=0x1)

  > ISDN1: msn='*' DNID='99546476' MSN
 == ISDN1: Incoming call '' -> '99546476'
INFO_IND ID=001 #0x03a7 LEN=0024
 Controller/PLCI/NCCI= 0x101
 InfoNumber  = 0x70
 InfoElement = 99546476

INFO_RESP ID=001 #0x03a7 LEN=0012
 Controller/PLCI/NCCI= 0x101

   -- ISDN1: info element CALLED PARTY NUMBER
  > ISDN1: INFO_IND DID digits not used in this state.
INFO_IND ID=001 #0x03a8 LEN=0016
 Controller/PLCI/NCCI= 0x101
 InfoNumber  = 0x18
 InfoElement = <89>

INFO_RESP ID=001 #0x03a8 LEN=0012
 Controller/PLCI/NCCI= 0x101

   -- ISDN1: info element CHANNEL IDENTIFICATION 89
INFO_IND ID=001 #0x03a9 LEN=0016
 Controller/PLCI/NCCI= 0x101
 InfoNumber  = 0xa1
 InfoElement = 

INFO_RESP ID=001 #0x03a9 LEN=0012
 Controller/PLCI/NCCI= 0x101

   -- ISDN1: info element Sending Complete
CONNECT_RESP ID=001 #0x03a9 LEN=0032
 Controller/PLCI/NCCI= 0x101
 Reject  = 0x1
 BProtocol
  B1protocol = 0x0
  B2protocol = 0x0
  B3protocol = 0x0
  B1configuration= default
  B2configuration= default
  B3configuration= default
 ConnectedNumber = default
 ConnectedSubaddress = default
 LLC = default
 AdditionalInfo
  BChannelinformation= default
  Keypadfacility = default
  Useruserdata   = default
  Facilitydataarray  = default

DISCONNECT_IND ID=001 #0x03aa LEN=0014
 Controller/PLCI/NCCI= 0x101
 Reason  = 0x0

DISCONNECT_RESP ID=001 #0x03aa LEN=0012
 Controller/PLCI/NCCI= 0x101

 == ISDN1: CAPI Hangingup
 == ISDN1: Interface cleanup PLCI=0x101

I will apreciate your assistance

Esteban

_
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http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Fninemsn%2Eseek%2Ecom%2Eau&_t=752315885&_r=Jan05_tagline&_m=EXT


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Re: [Asterisk-Users] chan_capi-cm and type of number problem (ToN)

2006-05-17 Thread Armin Schindler

On Wed, 17 May 2006, Klaus Darilion wrote:
> Armin Schindler wrote:
> > On Tue, 16 May 2006, Klaus Darilion wrote:
> > > Hi!
> > > 
> > > I have problems with the ToN configurations in chan_capi-cm. I
> > > understand how
> > > incoming calls are rewritten using national and international prefix.
> > > But for
> > > outgoing calls - what is the ToN?
> > 
> > I never really needed ton in TE mode, but when your card is in NT mode,
> > setting the ton may be important.
> > 
> > > Further, is there any debug info of the Ton? "capi debug" and the
> > > divactrl
> > > dchannel both show the CLI and CDLI, but do not show the ToN.
> > 
> > The ton is a byte prepended to the callerid and can be seen when looking
> > at the capi messages. On outgoing call when 'capi debug' is set, the
> > presentation and ton value is shown as well.
> 
> Hi Armin!
> 
> I think the capi debug is too verbose. When I debug, I'm mostly interested in
> D-channel info, not B-channel. What do you think about splitting this into
> "capi debug dchannel" and
> "capi debug"

on 'capi debug' you see messages for your set verbose level only.
With 'set verbose 5' you see the 'dchannel' stuff, with 'set verbose 9'
you see everything.
 
> > > How can I set the ToN? I suspect chan_capi to use the received ToN
> > > also for
> > > outgoing calls when bridging calls. How can I verify this?
> > 
> > When using a newer Asterisk with cid_ton (I think 1.2.x has them), then
> > chan-capi will set ton on incoming call and on outgoing call, this value
> > is used as well. So it is bridged.
> 
> This is IMO a strange thing, combined with the default behavior of national
> and internationalprefix.
> 
> e.g. a call is received with CLI=4912345 TON=international
> default internationalprefix=00

Yes, but that can be changed in capi.conf.
 
> Thus, chan_capi rewrites the number to 004912345, cid_ton is still
> international.
> 
> When this call is bridged to the PBX, it sends CLI=004912345 TON=international
> 
> Thus, if the number is rewritten, IMO also the context should be rewritten to
> UNKNOWN.

The user has two possibilities:
a) use the chan-capi feature of automatically prepend the prefixes according to 
TON
b) don't use automatic prefix setting in capi.conf and do your own stuff 
   according to CALLERTON in your dialplan

both should not be mixed, because (as you stated above) will cause double 
changes. a) is for standard usage (one ISDN port, no NT-mode), b) is the 
professional version.
 
> > Since Asterisk does not provide (as far as I know) a possibility to
> > change that cid_ton value, chan-capi will overwrite that value with the
> > value in
> > variable ${CALLERTON}, if set.
> 
> As I see this must be a number. Using strings (INTERNATIONAL, LOCAL, NATIONAL)
> would be nice too.

Hmm, you can device variables with that in your dialplan. Or Asterisk 
provides them...
 
> Not sure if I understand the source code right: CALLERTON will be read to set
> callers TON on outgoing calls, whereas CALLEDTON will be set on incoming
> called TON? Why not set both variables in incoming calls and read both
> variables on outgoing calls?

CALLEDTON is set by chan-capi for the TON of the called number, not the 
caller id! Normaly you don't need that TON and Asterisk doesn't provide
any feature for that.

The CALLERTON is the widely used TON and it is set to asterisks cid_ton 
internal variable, which can be read via ${CALLERTON}. But since Asterisk 
does not set cid_ton when writing to CALLERTON, chan-capi evaluates this
variable by itself (when set).

Armin
 
> regards
> Klaus
> 
> > When you use the latest chan-capi from trunk (HEAD version), you can use
> > 'capi show channels' to see the ton as well.
> > 
> > Armin
> > 
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Re: [Asterisk-Users] chan_capi-cm and dialing without number

2006-05-17 Thread Klaus Darilion


Armin Schindler wrote:

On Tue, 16 May 2006, Klaus Darilion wrote:

Does someone have any hints?

I would need a 'capi debug' log to say more, but chan-capi receives
the command for dial these digits twice.
Just a wild guess, but do you have softdtmf detection activated in
addition?

Yes - softdmtf is on! Thanks
for the hint.

btw: I use WaitExten to collect digits. Is softdtmf required for this
application?


softdtmf is needed when the hard-dtmf (DSP) is not available. If you use 
Eicon diva server cards, then softdtmf is never required.


Thanks. I've no set softdtmf and relaxdtmf to "off" and things work fine 
now.


regards
klaus
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Re: [Asterisk-Users] chan_capi-cm and type of number problem (ToN)

2006-05-17 Thread Klaus Darilion


Armin Schindler wrote:

On Tue, 16 May 2006, Klaus Darilion wrote:

Hi!

I have problems with the ToN configurations in chan_capi-cm. I understand how
incoming calls are rewritten using national and international prefix. But for
outgoing calls - what is the ToN?


I never really needed ton in TE mode, but when your card is in NT mode, 
setting the ton may be important.
 

Further, is there any debug info of the Ton? "capi debug" and the divactrl
dchannel both show the CLI and CDLI, but do not show the ToN.


The ton is a byte prepended to the callerid and can be seen when looking at 
the capi messages. On outgoing call when 'capi debug' is set, the

presentation and ton value is shown as well.


Hi Armin!

I think the capi debug is too verbose. When I debug, I'm mostly 
interested in D-channel info, not B-channel. What do you think about 
splitting this into

"capi debug dchannel" and
"capi debug"


How can I set the ToN? I suspect chan_capi to use the received ToN also for
outgoing calls when bridging calls. How can I verify this?


When using a newer Asterisk with cid_ton (I think 1.2.x has them), then 
chan-capi will set ton on incoming call and on outgoing call, this value is 
used as well. So it is bridged.


This is IMO a strange thing, combined with the default behavior of 
national and internationalprefix.


e.g. a call is received with CLI=4912345 TON=international
default internationalprefix=00

Thus, chan_capi rewrites the number to 004912345, cid_ton is still 
international.


When this call is bridged to the PBX, it sends CLI=004912345 
TON=international


Thus, if the number is rewritten, IMO also the context should be 
rewritten to UNKNOWN.


Since Asterisk does not provide (as far as I know) a possibility to change 
that cid_ton value, chan-capi will overwrite that value with the value in

variable ${CALLERTON}, if set.


As I see this must be a number. Using strings (INTERNATIONAL, LOCAL, 
NATIONAL) would be nice too.


Not sure if I understand the source code right: CALLERTON will be read 
to set callers TON on outgoing calls, whereas CALLEDTON will be set on 
incoming called TON? Why not set both variables in incoming calls and 
read both variables on outgoing calls?


regards
Klaus


When you use the latest chan-capi from trunk (HEAD version), you can use
'capi show channels' to see the ton as well.

Armin

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Re: [Asterisk-Users] chan_capi-cm and dialing without number

2006-05-16 Thread Armin Schindler

On Tue, 16 May 2006, Klaus Darilion wrote:
> Klaus Darilion wrote:
> > Armin Schindler wrote:
> > > On Tue, 16 May 2006, Klaus Darilion wrote:
> > > > Klaus Darilion wrote:
> > > > > Hi!
> > > > > 
> > > > > Is it possible to bridge a call (incoming call leg is from
> > > > > the PBX,
> > > > > outgoing call leg is to the PSTN) without even dialing a
> > > > > number. Thus,
> > > > > the dial tone received from the PSTN should be forwarded to
> > > > > the PBX, and
> > > > > the overlap digits received from the PBX should be forwarded
> > > > > to the PSTN.
> > > > > 
> > > > > I tried the following, but it does not work. Is there a
> > > > > workaround?
> > > > > 
> > > > > [frompbx]
> > > > > exten => s,1,Dial(CAPI/g1//b,90)
> > > > update: this works (i had a typo and use B isntead of b).
> > > > 
> > > > But the problem is that each digit received is sent twice to the
> > > > net:
> > > > 
> > > > Does someone have any hints?
> > > 
> > > I would need a 'capi debug' log to say more, but chan-capi receives
> > > the command for dial these digits twice.
> > > Just a wild guess, but do you have softdtmf detection activated in
> > > addition?
> > 
> > Yes - softdmtf is on! Thanks
> > for the hint.
> 
> btw: I use WaitExten to collect digits. Is softdtmf required for this
> application?

softdtmf is needed when the hard-dtmf (DSP) is not available. If you use 
Eicon diva server cards, then softdtmf is never required.

Armin
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Re: [Asterisk-Users] chan_capi-cm and dialing without number

2006-05-16 Thread Klaus Darilion


Klaus Darilion wrote:

Armin Schindler wrote:

On Tue, 16 May 2006, Klaus Darilion wrote:

Klaus Darilion wrote:

Hi!

Is it possible to bridge a call (incoming call leg is from the PBX,
outgoing call leg is to the PSTN) without even dialing a number. Thus,
the dial tone received from the PSTN should be forwarded to the PBX, 
and
the overlap digits received from the PBX should be forwarded to the 
PSTN.


I tried the following, but it does not work. Is there a workaround?

[frompbx]
exten => s,1,Dial(CAPI/g1//b,90)

update: this works (i had a typo and use B isntead of b).

But the problem is that each digit received is sent twice to the net:

Does someone have any hints?


I would need a 'capi debug' log to say more, but chan-capi receives 
the command for dial these digits twice.
Just a wild guess, but do you have softdtmf detection activated in 
addition?


Yes - softdmtf is on! Thanks
for the hint.


btw: I use WaitExten to collect digits. Is softdtmf required for this 
application?


regards
klaus
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Re: [Asterisk-Users] chan_capi-cm and dialing without number

2006-05-16 Thread Klaus Darilion


Armin Schindler wrote:

On Tue, 16 May 2006, Klaus Darilion wrote:

Klaus Darilion wrote:

Hi!

Is it possible to bridge a call (incoming call leg is from the PBX,
outgoing call leg is to the PSTN) without even dialing a number. Thus,
the dial tone received from the PSTN should be forwarded to the PBX, and
the overlap digits received from the PBX should be forwarded to the PSTN.

I tried the following, but it does not work. Is there a workaround?

[frompbx]
exten => s,1,Dial(CAPI/g1//b,90)

update: this works (i had a typo and use B isntead of b).

But the problem is that each digit received is sent twice to the net:

Does someone have any hints?


I would need a 'capi debug' log to say more, but chan-capi receives the 
command for dial these digits twice.

Just a wild guess, but do you have softdtmf detection activated in addition?


Yes - softdmtf is on! Thanks
for the hint.

Armin, thanks a lot for all the support
Klaus




Armin
 

thanks
klaus

-- Executing Goto("CAPI/ISDN4/-34", "fromfax|s|1") in new stack
-- Goto (fromfax,s,1)
-- Executing Dial("CAPI/ISDN4/-34", "CAPI/g1//b|90") in new stack
-- Called g1//b
-- CAPI/ISDN2/-35 is making progress passing it to CAPI/ISDN4/-34
  == ISDN2: Setting up echo canceller (PLCI=0x302, function=1, options=4,
tail=64)
  == ISDN4: Setting up echo canceller (PLCI=0x104, function=1, options=4,
tail=64)
-- ISDN2: Echo canceller successfully set up (PLCI=0x302)
-- ISDN4: Echo canceller successfully set up (PLCI=0x104)
-- ISDN4: Updated channel name: CAPI/ISDN4/0-36
-- ISDN2: Updated channel name: CAPI/ISDN2/0-37
-- ISDN2: Updated channel name: CAPI/ISDN2/00-38
-- ISDN4: Updated channel name: CAPI/ISDN4/00-39
-- ISDN2: Updated channel name: CAPI/ISDN2/000-3a
-- ISDN2: Updated channel name: CAPI/ISDN2/-3b
-- ISDN4: Updated channel name: CAPI/ISDN4/004-3c
-- ISDN2: Updated channel name: CAPI/ISDN2/4-3d
-- ISDN2: Updated channel name: CAPI/ISDN2/44-3e
-- ISDN4: Updated channel name: CAPI/ISDN4/0043-3f
-- ISDN2: Updated channel name: CAPI/ISDN2/443-40
-- ISDN2: Updated channel name: CAPI/ISDN2/4433-41
-- ISDN4: Updated channel name: CAPI/ISDN4/00437-42
-- ISDN2: Updated channel name: CAPI/ISDN2/44337-43
-- ISDN2: Updated channel name: CAPI/ISDN2/443377-44
-- ISDN4: Updated channel name: CAPI/ISDN4/004372-45
-- ISDN2: Updated channel name: CAPI/ISDN2/4433772-46
-- ISDN2: Updated channel name: CAPI/ISDN2/44337722-47
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Re: [Asterisk-Users] chan_capi-cm and dialing without number

2006-05-16 Thread Armin Schindler

On Tue, 16 May 2006, Klaus Darilion wrote:
> Klaus Darilion wrote:
> > Hi!
> > 
> > Is it possible to bridge a call (incoming call leg is from the PBX,
> > outgoing call leg is to the PSTN) without even dialing a number. Thus,
> > the dial tone received from the PSTN should be forwarded to the PBX, and
> > the overlap digits received from the PBX should be forwarded to the PSTN.
> > 
> > I tried the following, but it does not work. Is there a workaround?
> > 
> > [frompbx]
> > exten => s,1,Dial(CAPI/g1//b,90)
> 
> update: this works (i had a typo and use B isntead of b).
> 
> But the problem is that each digit received is sent twice to the net:
> 
> Does someone have any hints?

I would need a 'capi debug' log to say more, but chan-capi receives the 
command for dial these digits twice.
Just a wild guess, but do you have softdtmf detection activated in addition?

Armin
 
> thanks
> klaus
> 
> -- Executing Goto("CAPI/ISDN4/-34", "fromfax|s|1") in new stack
> -- Goto (fromfax,s,1)
> -- Executing Dial("CAPI/ISDN4/-34", "CAPI/g1//b|90") in new stack
> -- Called g1//b
> -- CAPI/ISDN2/-35 is making progress passing it to CAPI/ISDN4/-34
>   == ISDN2: Setting up echo canceller (PLCI=0x302, function=1, options=4,
> tail=64)
>   == ISDN4: Setting up echo canceller (PLCI=0x104, function=1, options=4,
> tail=64)
> -- ISDN2: Echo canceller successfully set up (PLCI=0x302)
> -- ISDN4: Echo canceller successfully set up (PLCI=0x104)
> -- ISDN4: Updated channel name: CAPI/ISDN4/0-36
> -- ISDN2: Updated channel name: CAPI/ISDN2/0-37
> -- ISDN2: Updated channel name: CAPI/ISDN2/00-38
> -- ISDN4: Updated channel name: CAPI/ISDN4/00-39
> -- ISDN2: Updated channel name: CAPI/ISDN2/000-3a
> -- ISDN2: Updated channel name: CAPI/ISDN2/-3b
> -- ISDN4: Updated channel name: CAPI/ISDN4/004-3c
> -- ISDN2: Updated channel name: CAPI/ISDN2/4-3d
> -- ISDN2: Updated channel name: CAPI/ISDN2/44-3e
> -- ISDN4: Updated channel name: CAPI/ISDN4/0043-3f
> -- ISDN2: Updated channel name: CAPI/ISDN2/443-40
> -- ISDN2: Updated channel name: CAPI/ISDN2/4433-41
> -- ISDN4: Updated channel name: CAPI/ISDN4/00437-42
> -- ISDN2: Updated channel name: CAPI/ISDN2/44337-43
> -- ISDN2: Updated channel name: CAPI/ISDN2/443377-44
> -- ISDN4: Updated channel name: CAPI/ISDN4/004372-45
> -- ISDN2: Updated channel name: CAPI/ISDN2/4433772-46
> -- ISDN2: Updated channel name: CAPI/ISDN2/44337722-47
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Re: [Asterisk-Users] chan_capi-cm and dialing without number

2006-05-16 Thread Armin Schindler

On Tue, 16 May 2006, Klaus Darilion wrote:
> Hi!
> 
> Is it possible to bridge a call (incoming call leg is from the PBX, outgoing
> call leg is to the PSTN) without even dialing a number. Thus, the dial tone
> received from the PSTN should be forwarded to the PBX, and the overlap digits
> received from the PBX should be forwarded to the PSTN.

Yes, this is possible. I use that all the time.
 
> I tried the following, but it does not work. Is there a workaround?
> 
> [frompbx]
> exten => s,1,Dial(CAPI/g1//b,90)

That is correct. I use options /bo , but that should not make any difference 
in that case. At least that example must provide a dialtone to you and the 
possibility to dial your digits.

What exactly does not work?

Armin

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Re: [Asterisk-Users] chan_capi-cm and type of number problem (ToN)

2006-05-16 Thread Armin Schindler

On Tue, 16 May 2006, Klaus Darilion wrote:
> Hi!
> 
> I have problems with the ToN configurations in chan_capi-cm. I understand how
> incoming calls are rewritten using national and international prefix. But for
> outgoing calls - what is the ToN?

I never really needed ton in TE mode, but when your card is in NT mode, 
setting the ton may be important.

> Further, is there any debug info of the Ton? "capi debug" and the divactrl
> dchannel both show the CLI and CDLI, but do not show the ToN.

The ton is a byte prepended to the callerid and can be seen when looking at 
the capi messages. On outgoing call when 'capi debug' is set, the
presentation and ton value is shown as well.

> How can I set the ToN? I suspect chan_capi to use the received ToN also for
> outgoing calls when bridging calls. How can I verify this?

When using a newer Asterisk with cid_ton (I think 1.2.x has them), then 
chan-capi will set ton on incoming call and on outgoing call, this value is 
used as well. So it is bridged.
Since Asterisk does not provide (as far as I know) a possibility to change 
that cid_ton value, chan-capi will overwrite that value with the value in
variable ${CALLERTON}, if set.
When you use the latest chan-capi from trunk (HEAD version), you can use
'capi show channels' to see the ton as well.

Armin

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Re: [Asterisk-Users] chan_capi-cm and dialing without number

2006-05-16 Thread Klaus Darilion


Klaus Darilion wrote:

Hi!

Is it possible to bridge a call (incoming call leg is from the PBX, 
outgoing call leg is to the PSTN) without even dialing a number. Thus, 
the dial tone received from the PSTN should be forwarded to the PBX, and 
the overlap digits received from the PBX should be forwarded to the PSTN.


I tried the following, but it does not work. Is there a workaround?

[frompbx]
exten => s,1,Dial(CAPI/g1//b,90)


update: this works (i had a typo and use B isntead of b).

But the problem is that each digit received is sent twice to the net:

Does someone have any hints?

thanks
klaus

   -- Executing Goto("CAPI/ISDN4/-34", "fromfax|s|1") in new stack
-- Goto (fromfax,s,1)
-- Executing Dial("CAPI/ISDN4/-34", "CAPI/g1//b|90") in new stack
-- Called g1//b
-- CAPI/ISDN2/-35 is making progress passing it to CAPI/ISDN4/-34
  == ISDN2: Setting up echo canceller (PLCI=0x302, function=1, 
options=4, tail=64)
  == ISDN4: Setting up echo canceller (PLCI=0x104, function=1, 
options=4, tail=64)

-- ISDN2: Echo canceller successfully set up (PLCI=0x302)
-- ISDN4: Echo canceller successfully set up (PLCI=0x104)
-- ISDN4: Updated channel name: CAPI/ISDN4/0-36
-- ISDN2: Updated channel name: CAPI/ISDN2/0-37
-- ISDN2: Updated channel name: CAPI/ISDN2/00-38
-- ISDN4: Updated channel name: CAPI/ISDN4/00-39
-- ISDN2: Updated channel name: CAPI/ISDN2/000-3a
-- ISDN2: Updated channel name: CAPI/ISDN2/-3b
-- ISDN4: Updated channel name: CAPI/ISDN4/004-3c
-- ISDN2: Updated channel name: CAPI/ISDN2/4-3d
-- ISDN2: Updated channel name: CAPI/ISDN2/44-3e
-- ISDN4: Updated channel name: CAPI/ISDN4/0043-3f
-- ISDN2: Updated channel name: CAPI/ISDN2/443-40
-- ISDN2: Updated channel name: CAPI/ISDN2/4433-41
-- ISDN4: Updated channel name: CAPI/ISDN4/00437-42
-- ISDN2: Updated channel name: CAPI/ISDN2/44337-43
-- ISDN2: Updated channel name: CAPI/ISDN2/443377-44
-- ISDN4: Updated channel name: CAPI/ISDN4/004372-45
-- ISDN2: Updated channel name: CAPI/ISDN2/4433772-46
-- ISDN2: Updated channel name: CAPI/ISDN2/44337722-47
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[Asterisk-Users] chan_capi-cm and dialing without number

2006-05-16 Thread Klaus Darilion


Hi!

Is it possible to bridge a call (incoming call leg is from the PBX, 
outgoing call leg is to the PSTN) without even dialing a number. Thus, 
the dial tone received from the PSTN should be forwarded to the PBX, and 
the overlap digits received from the PBX should be forwarded to the PSTN.


I tried the following, but it does not work. Is there a workaround?

[frompbx]
exten => s,1,Dial(CAPI/g1//b,90)

thanks
klaus
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[Asterisk-Users] chan_capi-cm and type of number problem (ToN)

2006-05-16 Thread Klaus Darilion


Hi!

I have problems with the ToN configurations in chan_capi-cm. I 
understand how incoming calls are rewritten using national and 
international prefix. But for outgoing calls - what is the ToN?


Further, is there any debug info of the Ton? "capi debug" and the 
divactrl dchannel both show the CLI and CDLI, but do not show the ToN.


How can I set the ToN? I suspect chan_capi to use the received ToN also 
for outgoing calls when bridging calls. How can I verify this?


thanks
Klaus
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RE: [Asterisk-Users] chan_capi and Eicon Diva

2006-02-28 Thread David Waugh

Hello Paolo,

I put together this page which has instructions on getting Asterisk
working with a Diva Server card. Follow the steps for Option 0...

http://www.voip-info.org/wiki-Asterisk+Eicon+Diva+CAPI+ISDN

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paolo
Prandini
Sent: 28 February 2006 07:28
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] chan_capi and Eicon Diva

I am trying to use chan_capi with an Eicon Diva Server BRI.
I installed the Eicon drivers from source including CAPU and
I can use the board correcly using tty_test and minicom over
/dev/ttyds01
or /dev/ttyds01.
I need to insmod capi ( why? it is not written anywhere) and then
capiinfo shows me all board parameters, but I have to break it otherwise
capiinfo doesn't exit at all, but I don't know if this is an expected
behaviour.
When I try to use chan_capi I get the message in the asterisk log that
CAPI is not installed and in fact the capi20_isinstalled function in
chan_capi.c returns 4109, the error code expected when capi is not
installed.
Why? Has anyone some experience on the matter that is willing to share?
I have looked everywhere on google and the usual forums but there are
no useful informations.
Thanks.
Paolo
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Re: [Asterisk-Users] chan_capi and Eicon Diva

2006-02-28 Thread Armin Schindler

On Tue, 28 Feb 2006, Paolo Prandini wrote:
> I am trying to use chan_capi with an Eicon Diva Server BRI.
> I installed the Eicon drivers from source including CAPU and
> I can use the board correcly using tty_test and minicom over /dev/ttyds01
> or /dev/ttyds01.
> I need to insmod capi ( why? it is not written anywhere) and then

If not already loaded, of course, to use CAPI to need to insmod the modules
for that feature. The main module is kernelcapi, which is needed by the
divacapi module. the module 'capi' provides the user-space access via
/dev/capi20

> capiinfo shows me all board parameters, but I have to break it otherwise
> capiinfo doesn't exit at all, but I don't know if this is an expected
> behaviour.

No, capiinfo may not wait, it should exit immediatly.

> When I try to use chan_capi I get the message in the asterisk log that
> CAPI is not installed and in fact the capi20_isinstalled function in
> chan_capi.c returns 4109, the error code expected when capi is not
> installed.
> Why? Has anyone some experience on the matter that is willing to share?
> I have looked everywhere on google and the usual forums but there are
> no useful informations.

Does /dev/capi20 has the correct permissions set for asterisk?

Armin
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[Asterisk-Users] chan_capi and Eicon Diva

2006-02-27 Thread Paolo Prandini


I am trying to use chan_capi with an Eicon Diva Server BRI.
I installed the Eicon drivers from source including CAPU and
I can use the board correcly using tty_test and minicom over /dev/ttyds01
or /dev/ttyds01.
I need to insmod capi ( why? it is not written anywhere) and then
capiinfo shows me all board parameters, but I have to break it otherwise
capiinfo doesn't exit at all, but I don't know if this is an expected
behaviour.
When I try to use chan_capi I get the message in the asterisk log that
CAPI is not installed and in fact the capi20_isinstalled function in
chan_capi.c returns 4109, the error code expected when capi is not
installed.
Why? Has anyone some experience on the matter that is willing to share?
I have looked everywhere on google and the usual forums but there are
no useful informations.
Thanks.
Paolo
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Re: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem

2006-02-24 Thread Faris Raouf


Thank you Armin. This is extremely useful.

Faris.


Armin Schindler wrote:
There are three possibilities to set the CallingPartyNumber (own number for 
outgoing):


1) Set(CALLERID(number)=12345)
   before Dial()

2) Dial(CAPI/contr1/12345:${EXTEN}/)

3) Dial(CAPI/contr1/${EXTEN}/d,...) and 'defaultcid=12345' in capi.conf
   with this defaultcid you can set a number for each interface in capi.conf
   and activate that by the /d option. This is useful when you combined more 
   than one interface into one group, but need to use a correct (and 
   different) number on dialout with e.g. 'g1', because the dialplan 
   doesn't know which interface will be used.


Armin

On Thu, 23 Feb 2006, Faris Raouf wrote:

Thanks for that Peter!

I think your message solved my problem: I set the master number to be in group
1 (group=1) in capi.conf and called Dial with CAPI/g1 and it worked perfectly.

However, with group=1 in capi.conf for the master number, at the moment no
matter what I do I'm getting the master number presented as the CLI. This is
fine by me because it is exactly what I want, but it is all very confusing :-)

Faris.


Peter Braidwood wrote:

I have ISDN2 and 10 MSN numbers and also have just upgraded to 1.2.4 and
chan_capi-cm and have this working completely perfectly

Capi.conf

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=en

[ISDN1]
isdnmode=msn
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=from-isdn
group=1
devices=2

bit of extensions.conf, I dial 9 for an outside line

[pstn]

exten => _9./321,1,Dial(CAPI/g1/01234567890:${EXTEN:1})
exten => _9./322,1,Dial(CAPI/g1/01234567891:${EXTEN:1})
exten => _9./323,1,Dial(CAPI/g1/01234567892:${EXTEN:1})
exten => _9./324,1,Dial(CAPI/g1/01234567893:${EXTEN:1})
exten => _9./326,1,Dial(CAPI/g1/01234567894:${EXTEN:1})
exten => _9./327,1,Dial(CAPI/g1/01234567895:${EXTEN:1})
exten => _9./328,1,Dial(CAPI/g1/01234567896:${EXTEN:1})
exten => _9./350,1,Dial(CAPI/g1/01234567897:${EXTEN:1})
exten => _9./351,1,Dial(CAPI/g1/01234567898:${EXTEN:1})
exten => _9./352,1,Dial(CAPI/g1/01234567899:${EXTEN:1})

So when extension 326 dials out the cli that is presented would be
01234567894

Contact me off list if you want any further help.

Peter Braidwood


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: 23 February 2006 13:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem

I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm
0.6.4

When making outgoing calls I don't seem to have any control over the CLI

that is presented to the called party -- it can be any one of the MSNs
allocated to the line, allocated on what seems to be a random basis.

This is on a BT Business Highway line (which is essentially an ISDN2e
line with two built-in analog ports), configured with 8MSNs alongside the
single the "master" digital telephone number for the line.

With a much older version of chan_capi-cm (0.5.4 I think) and Asterisk
1.0.9 it was always the "master" number that was presented, and that is
actually what I want.

Obviously the format of capi.conf has changed between these two versions

of chan_capi-cm, so maybe I'm doing something wrong. Any suggestions
would be appreciated.

Here's my capi.conf (actual numbers changed to protect the innocent!)

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
; ulaw=yes;set this, if you live in u-law world instead of
; a-law

[123456]
;  Master number for line - i.e. number for line before MSNs were
allocated
;  and which still works and still accepts incoming calls.
isdnmode=msn
msn=01234123456
; incomingmsn=*
incomingmsn=123456
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
; echotail=64
; callgroup=1
; deflect=12345678
devices=2

[123457]
; first MSN
msn=01234123457
; incomingmsn=*
incomingmsn=123457
isdnmode=msn
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
; echotail=64
; callgroup=1
; deflect=12345678
devices=2

{repeated for next 7 MSNs}


And in extensions.conf I have:

[globals]
ISDN1=CAPI/123456


[mysip]

; GET OUTSIDE LINE (ISDN1 - dial 9)
ignorepat => 9
exten => exten => _9.,1,Dial(${ISDN1}/${EXTEN:1}/b)
exten => _9.,2,Playback(busy)
exten => _9.,3,Hangup

*

I've tried using ISDN1=CAPI/contr1 but it makes no difference.
I've tried leaving out the isdnmode=msn but it makes no difference
I've tried entering 01234123456 as the msn= line on all of the msn
entries in capi.conf but it makes no difference either.

And now I'm out of ideas and any help would be appreciated.

Thanks,

Faris.

p.s. sorry if this message is HTML. I've switched to using Thunde

Re: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem

2006-02-24 Thread Armin Schindler

There are three possibilities to set the CallingPartyNumber (own number for 
outgoing):

1) Set(CALLERID(number)=12345)
   before Dial()

2) Dial(CAPI/contr1/12345:${EXTEN}/)

3) Dial(CAPI/contr1/${EXTEN}/d,...) and 'defaultcid=12345' in capi.conf
   with this defaultcid you can set a number for each interface in capi.conf
   and activate that by the /d option. This is useful when you combined more 
   than one interface into one group, but need to use a correct (and 
   different) number on dialout with e.g. 'g1', because the dialplan 
   doesn't know which interface will be used.

Armin

On Thu, 23 Feb 2006, Faris Raouf wrote:
> Thanks for that Peter!
> 
> I think your message solved my problem: I set the master number to be in group
> 1 (group=1) in capi.conf and called Dial with CAPI/g1 and it worked perfectly.
> 
> However, with group=1 in capi.conf for the master number, at the moment no
> matter what I do I'm getting the master number presented as the CLI. This is
> fine by me because it is exactly what I want, but it is all very confusing :-)
> 
> Faris.
> 
> 
> Peter Braidwood wrote:
> > I have ISDN2 and 10 MSN numbers and also have just upgraded to 1.2.4 and
> > chan_capi-cm and have this working completely perfectly
> > 
> > Capi.conf
> > 
> > [general]
> > nationalprefix=0
> > internationalprefix=00
> > rxgain=0.8
> > txgain=0.8
> > language=en
> > 
> > [ISDN1]
> > isdnmode=msn
> > incomingmsn=*
> > controller=1
> > softdtmf=1
> > accountcode=
> > context=from-isdn
> > group=1
> > devices=2
> > 
> > bit of extensions.conf, I dial 9 for an outside line
> > 
> > [pstn]
> > 
> > exten => _9./321,1,Dial(CAPI/g1/01234567890:${EXTEN:1})
> > exten => _9./322,1,Dial(CAPI/g1/01234567891:${EXTEN:1})
> > exten => _9./323,1,Dial(CAPI/g1/01234567892:${EXTEN:1})
> > exten => _9./324,1,Dial(CAPI/g1/01234567893:${EXTEN:1})
> > exten => _9./326,1,Dial(CAPI/g1/01234567894:${EXTEN:1})
> > exten => _9./327,1,Dial(CAPI/g1/01234567895:${EXTEN:1})
> > exten => _9./328,1,Dial(CAPI/g1/01234567896:${EXTEN:1})
> > exten => _9./350,1,Dial(CAPI/g1/01234567897:${EXTEN:1})
> > exten => _9./351,1,Dial(CAPI/g1/01234567898:${EXTEN:1})
> > exten => _9./352,1,Dial(CAPI/g1/01234567899:${EXTEN:1})
> > 
> > So when extension 326 dials out the cli that is presented would be
> > 01234567894
> > 
> > Contact me off list if you want any further help.
> > 
> > Peter Braidwood
> > 
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Faris
> > Raouf
> > Sent: 23 February 2006 13:24
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem
> > 
> > I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm
> > 0.6.4
> > 
> > When making outgoing calls I don't seem to have any control over the CLI
> > 
> > that is presented to the called party -- it can be any one of the MSNs
> > allocated to the line, allocated on what seems to be a random basis.
> > 
> > This is on a BT Business Highway line (which is essentially an ISDN2e
> > line with two built-in analog ports), configured with 8MSNs alongside the
> > single the "master" digital telephone number for the line.
> > 
> > With a much older version of chan_capi-cm (0.5.4 I think) and Asterisk
> > 1.0.9 it was always the "master" number that was presented, and that is
> > actually what I want.
> > 
> > Obviously the format of capi.conf has changed between these two versions
> > 
> > of chan_capi-cm, so maybe I'm doing something wrong. Any suggestions
> > would be appreciated.
> > 
> > Here's my capi.conf (actual numbers changed to protect the innocent!)
> > 
> > [general]
> > nationalprefix=0
> > internationalprefix=00
> > rxgain=0.8
> > txgain=0.8
> > ; ulaw=yes;set this, if you live in u-law world instead of
> > ; a-law
> > 
> > [123456]
> > ;  Master number for line - i.e. number for line before MSNs were
> > allocated
> > ;  and which still works and still accepts incoming calls.
> > isdnmode=msn
> > msn=01234123456
> > ; incomingmsn=*
> > incomingmsn=123456
> > controller=1
> > softdtmf=1
> > accountcode=
> > context=isdn-in
> > echosquelch=1
> > echocancel=yes
> > ; echotail=64
> > ; callgroup=1

Re: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem

2006-02-23 Thread Faris Raouf


Thanks for that Peter!

I think your message solved my problem: I set the master number to be in 
group 1 (group=1) in capi.conf and called Dial with CAPI/g1 and it 
worked perfectly.


However, with group=1 in capi.conf for the master number, at the moment 
no matter what I do I'm getting the master number presented as the CLI. 
This is fine by me because it is exactly what I want, but it is all very 
confusing :-)


Faris.


Peter Braidwood wrote:

I have ISDN2 and 10 MSN numbers and also have just upgraded to 1.2.4 and
chan_capi-cm and have this working completely perfectly

Capi.conf

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=en

[ISDN1]
isdnmode=msn
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=from-isdn
group=1
devices=2

bit of extensions.conf, I dial 9 for an outside line

[pstn]

exten => _9./321,1,Dial(CAPI/g1/01234567890:${EXTEN:1})
exten => _9./322,1,Dial(CAPI/g1/01234567891:${EXTEN:1})
exten => _9./323,1,Dial(CAPI/g1/01234567892:${EXTEN:1})
exten => _9./324,1,Dial(CAPI/g1/01234567893:${EXTEN:1})
exten => _9./326,1,Dial(CAPI/g1/01234567894:${EXTEN:1})
exten => _9./327,1,Dial(CAPI/g1/01234567895:${EXTEN:1})
exten => _9./328,1,Dial(CAPI/g1/01234567896:${EXTEN:1})
exten => _9./350,1,Dial(CAPI/g1/01234567897:${EXTEN:1})
exten => _9./351,1,Dial(CAPI/g1/01234567898:${EXTEN:1})
exten => _9./352,1,Dial(CAPI/g1/01234567899:${EXTEN:1})

So when extension 326 dials out the cli that is presented would be
01234567894

Contact me off list if you want any further help.

Peter Braidwood


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: 23 February 2006 13:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem

I've having a big problem after having upgraded to 1.2.4 and 
chan_capi-cm 0.6.4


When making outgoing calls I don't seem to have any control over the CLI

that is presented to the called party -- it can be any one of the MSNs 
allocated to the line, allocated on what seems to be a random basis.


This is on a BT Business Highway line (which is essentially an ISDN2e 
line with two built-in analog ports), configured with 8MSNs alongside 
the single the "master" digital telephone number for the line.


With a much older version of chan_capi-cm (0.5.4 I think) and Asterisk 
1.0.9 it was always the "master" number that was presented, and that is 
actually what I want.


Obviously the format of capi.conf has changed between these two versions

of chan_capi-cm, so maybe I'm doing something wrong. Any suggestions 
would be appreciated.


Here's my capi.conf (actual numbers changed to protect the innocent!)

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
;ulaw=yes;set this, if you live in u-law world instead of a-law

[123456]
; Master number for line - i.e. number for line before MSNs were
allocated
; and which still works and still accepts incoming calls.
isdnmode=msn
msn=01234123456
;incomingmsn=*
incomingmsn=123456
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2

[123457]
;first MSN
msn=01234123457
;incomingmsn=*
incomingmsn=123457
isdnmode=msn
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2

{repeated for next 7 MSNs}


And in extensions.conf I have:

[globals]
ISDN1=CAPI/123456


[mysip]

;GET OUTSIDE LINE (ISDN1 - dial 9)
ignorepat => 9
exten => exten => _9.,1,Dial(${ISDN1}/${EXTEN:1}/b)
exten => _9.,2,Playback(busy)
exten => _9.,3,Hangup

*

I've tried using ISDN1=CAPI/contr1 but it makes no difference.
I've tried leaving out the isdnmode=msn but it makes no difference
I've tried entering 01234123456 as the msn= line on all of the msn 
entries in capi.conf but it makes no difference either.


And now I'm out of ideas and any help would be appreciated.

Thanks,

Faris.

p.s. sorry if this message is HTML. I've switched to using Thunderbird 
and it is confusing the heck out of me. It claims this is a plain text 
message but it doesn't look like plain text to me from this end!







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RE: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem

2006-02-23 Thread Peter Braidwood

I have ISDN2 and 10 MSN numbers and also have just upgraded to 1.2.4 and
chan_capi-cm and have this working completely perfectly

Capi.conf

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=en

[ISDN1]
isdnmode=msn
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=from-isdn
group=1
devices=2

bit of extensions.conf, I dial 9 for an outside line

[pstn]

exten => _9./321,1,Dial(CAPI/g1/01234567890:${EXTEN:1})
exten => _9./322,1,Dial(CAPI/g1/01234567891:${EXTEN:1})
exten => _9./323,1,Dial(CAPI/g1/01234567892:${EXTEN:1})
exten => _9./324,1,Dial(CAPI/g1/01234567893:${EXTEN:1})
exten => _9./326,1,Dial(CAPI/g1/01234567894:${EXTEN:1})
exten => _9./327,1,Dial(CAPI/g1/01234567895:${EXTEN:1})
exten => _9./328,1,Dial(CAPI/g1/01234567896:${EXTEN:1})
exten => _9./350,1,Dial(CAPI/g1/01234567897:${EXTEN:1})
exten => _9./351,1,Dial(CAPI/g1/01234567898:${EXTEN:1})
exten => _9./352,1,Dial(CAPI/g1/01234567899:${EXTEN:1})

So when extension 326 dials out the cli that is presented would be
01234567894

Contact me off list if you want any further help.

Peter Braidwood


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: 23 February 2006 13:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem

I've having a big problem after having upgraded to 1.2.4 and 
chan_capi-cm 0.6.4

When making outgoing calls I don't seem to have any control over the CLI

that is presented to the called party -- it can be any one of the MSNs 
allocated to the line, allocated on what seems to be a random basis.

This is on a BT Business Highway line (which is essentially an ISDN2e 
line with two built-in analog ports), configured with 8MSNs alongside 
the single the "master" digital telephone number for the line.

With a much older version of chan_capi-cm (0.5.4 I think) and Asterisk 
1.0.9 it was always the "master" number that was presented, and that is 
actually what I want.

Obviously the format of capi.conf has changed between these two versions

of chan_capi-cm, so maybe I'm doing something wrong. Any suggestions 
would be appreciated.

Here's my capi.conf (actual numbers changed to protect the innocent!)

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
;ulaw=yes;set this, if you live in u-law world instead of a-law

[123456]
; Master number for line - i.e. number for line before MSNs were
allocated
; and which still works and still accepts incoming calls.
isdnmode=msn
msn=01234123456
;incomingmsn=*
incomingmsn=123456
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2

[123457]
;first MSN
msn=01234123457
;incomingmsn=*
incomingmsn=123457
isdnmode=msn
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2

{repeated for next 7 MSNs}


And in extensions.conf I have:

[globals]
ISDN1=CAPI/123456


[mysip]

;GET OUTSIDE LINE (ISDN1 - dial 9)
ignorepat => 9
exten => exten => _9.,1,Dial(${ISDN1}/${EXTEN:1}/b)
exten => _9.,2,Playback(busy)
exten => _9.,3,Hangup

*

I've tried using ISDN1=CAPI/contr1 but it makes no difference.
I've tried leaving out the isdnmode=msn but it makes no difference
I've tried entering 01234123456 as the msn= line on all of the msn 
entries in capi.conf but it makes no difference either.

And now I'm out of ideas and any help would be appreciated.

Thanks,

Faris.

p.s. sorry if this message is HTML. I've switched to using Thunderbird 
and it is confusing the heck out of me. It claims this is a plain text 
message but it doesn't look like plain text to me from this end!


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[Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem

2006-02-23 Thread Faris Raouf

I've having a big problem after having upgraded to 1.2.4 and 
chan_capi-cm 0.6.4


When making outgoing calls I don't seem to have any control over the CLI 
that is presented to the called party -- it can be any one of the MSNs 
allocated to the line, allocated on what seems to be a random basis.


This is on a BT Business Highway line (which is essentially an ISDN2e 
line with two built-in analog ports), configured with 8MSNs alongside 
the single the "master" digital telephone number for the line.


With a much older version of chan_capi-cm (0.5.4 I think) and Asterisk 
1.0.9 it was always the "master" number that was presented, and that is 
actually what I want.


Obviously the format of capi.conf has changed between these two versions 
of chan_capi-cm, so maybe I'm doing something wrong. Any suggestions 
would be appreciated.


Here's my capi.conf (actual numbers changed to protect the innocent!)

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
;ulaw=yes;set this, if you live in u-law world instead of a-law

[123456]
; Master number for line - i.e. number for line before MSNs were allocated
; and which still works and still accepts incoming calls.
isdnmode=msn
msn=01234123456
;incomingmsn=*
incomingmsn=123456
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2

[123457]
;first MSN
msn=01234123457
;incomingmsn=*
incomingmsn=123457
isdnmode=msn
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2

{repeated for next 7 MSNs}


And in extensions.conf I have:

[globals]
ISDN1=CAPI/123456


[mysip]

;GET OUTSIDE LINE (ISDN1 - dial 9)
ignorepat => 9
exten => exten => _9.,1,Dial(${ISDN1}/${EXTEN:1}/b)
exten => _9.,2,Playback(busy)
exten => _9.,3,Hangup

*

I've tried using ISDN1=CAPI/contr1 but it makes no difference.
I've tried leaving out the isdnmode=msn but it makes no difference
I've tried entering 01234123456 as the msn= line on all of the msn 
entries in capi.conf but it makes no difference either.


And now I'm out of ideas and any help would be appreciated.

Thanks,

Faris.

p.s. sorry if this message is HTML. I've switched to using Thunderbird 
and it is confusing the heck out of me. It claims this is a plain text 
message but it doesn't look like plain text to me from this end!



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[Asterisk-Users] chan_capi-cm-0.6.4

2006-02-22 Thread Ralf Schlatterbeck

Hello Armin, hello List
I'm trying to get chan_capi working with asterisk from debian stable
(asterisk 1.0.7, the debian version number is 1:1.0.7.dfsg.1-2).
I managed to get it compiled by providing my own version of
ast_copy_string.
This is an Austrian PTP line.  I can do outgoing calls fine (no
comprehensive tests yet).  For incoming calls, I'm getting "No answer"
on the remote end (GSM-phone) and the last output on the asterisk
console (capi debug + set verbose 15) is
  == ISDN1: Incoming call '0650621' -> '11'

When I unplug the ISDN line *after several minutes* I'm getting a
disconnect indication with a layer 1 error.

Interesting is, that I receive an INFO_IND *before* the CONNECT_IND.
This looks like an interesting variation of Austrian ISDN to me.
I've tried both, immediate=no and immediate=yes in the config-file.
The attached log is with immediate=yes.
The behaviour is the similar to chan_capi-0.4 (I'm still using a
patched 0.35 version in my production system) and previous versions, for
these I had to apply a patch to make incoming calls work:
--- chan_capi-0.4.0-PRE1/chan_capi.c2005-05-09 20:45:02.0 +0200
+++ chan_capi-0.4.0-PRE1-hacked-rsc/chan_capi.c 2005-06-04 13:30:19.0 +0
@@ -2141,7 +2175,8 @@
ast_pthread_mutex_init(&(p->lock),NULL);
i->mypipe = p;
if (i->isdnmode) {
-   p->c = capi_new(i,AST_STATE_DOWN);
+//RSC
+   p->c = capi_new(i,AST_STATE_RING);
i->state = CAPI_STATE_DID;
} else {
p->c = capi_new(i,AST_STATE_RING);

This patch immediately hands the call to asterisk instead of waiting for
further protocol information that apparently never comes (the info_ind
that comes *before* the connect_ind?).


Other system data:
- AMD sempron running Kernel version 2.6.15.4
- mISDN mqueue from 2006-02-17
- Billion HFC card using hfcpci

snippet from /etc/modprobe.d/capi:
--->
alias /dev/capi20 hfcpci
alias char-major-68-0 hfcpci

install hfcpci /sbin/modprobe capi debug=1; \
/sbin/modprobe mISDN_core debug=1; \
/sbin/modprobe mISDN_l1 debug=1; \
/sbin/modprobe mISDN_l2 debug=1; \
/sbin/modprobe l3udss1 debug=1; \
/sbin/modprobe mISDN_capi debug=1; \
/sbin/modprobe mISDN_x25dte debug=0; \
/sbin/modprobe mISDN_dsp; \
/sbin/modprobe mISDN_dtmf; \
/sbin/modprobe --ignore-install hfcpci protocol=0x22 debug=1
<---


I'm attaching the full verbose 15 + capi debug log.

Ralf
-- 
Ralf Schlatterbeck
email: [EMAIL PROTECTED] FAX: +43/2243/26465/23

Connected to Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k currently running on fox (pid 
= 6069)
Verbosity is at least 15
fox*CLI> capi debug
CAPI Debugging Enabled
-- Remote UNIX connection
-- Starting simple switch on 'Zap/2-1'
-- Hungup 'Zap/2-1'
INFO_IND ID=001 #0x0748 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = <89>

INFO_RESP ID=001 #0x0748 LEN=0012
  Controller/PLCI/NCCI= 0x101

CAPI: INFO_IND no interface for PLCI=0x101
   > CAPI: Command=INFO_IND,0x8492: no interface for PLCI=0x101, 
MSGNUM=0x748!
CONNECT_IND ID=001 #0x0749 LEN=0046
  Controller/PLCI/NCCI= 0x101
  CIPValue= 0x10
  CalledPartyNumber   = <81>11
  CallingPartyNumber  = <21 83>650621
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BC  = <80 90 a3>
  LLC = default
  HLC = <91 81>
  AdditionalInfo 
   BChannelinformation= default
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default

-- CONNECT_IND (PLCI=0x101,DID=11,CID=650621,CIP=0x10,CONTROLLER=0x1)
   > ISDN1: msn='*' DNID='11' DID
  == ISDN1: Incoming call '0650621' -> '11'
-- Remote UNIX connection disconnected
DISCONNECT_IND ID=001 #0x074a LEN=0014
  Controller/PLCI/NCCI= 0x101
  Reason  = 0x3301

DISCONNECT_RESP ID=001 #0x074a LEN=0012
  Controller/PLCI/NCCI= 0x101

   > CAPI INFO 0x3301: Protocol error layer 1 (broken line or B-channel 
removed by signalling protocol)
-- ISDN1: DISCONNECT_IND on incoming without pbx, doing hangup.
  == ISDN1: CAPI Hangingup
  == ISDN1: Interface cleanup PLCI=0x101
fox*CLI> 

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Re: [Asterisk-Users] chan_capi setting ${DNIS}

2006-02-20 Thread Nathan Alberti



On 20/02/2006, at 12:08 PM, Andrew Furey wrote:


On 2/20/06, Nathan Alberti <[EMAIL PROTECTED]> wrote:

Is there a reason the variable ${DNIS} does not get set with incoming
calls via chan_capi ?

Is it related to the MSN=X in capi.conf ?


Just a guess, are you thinking of ${DNID} instead? There's no direct
mention of ${DNIS} on the wiki variables page, but ${DNID} works for
me with a BRI...

Andrew


Thanks Andrew, DNID was what I was meant.

Nathan.

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Re: [Asterisk-Users] chan_capi setting ${DNIS}

2006-02-20 Thread Armin Schindler

On Mon, 20 Feb 2006, Nathan Alberti wrote:
> Is there a reason the variable ${DNIS} does not get set with incoming calls
> via chan_capi ?

I don't know any channel setting DNIS. What are you expecting with that 
variable?
 
> Is it related to the MSN=X in capi.conf ?

No. msn= is obsolete and does not exist in chan_capi since many releases. 
 
> version = chan_capi-cm-0.6.3
> 
> example;
> 
> exten => _9555XX,1,NoOp, ${EXTEN}, ${DNIS}
 
If you mean the caller number, then try ${CALLERID(number)} 
 
Armin
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Re: [Asterisk-Users] chan_capi setting ${DNIS}

2006-02-19 Thread Andrew Furey

On 2/20/06, Nathan Alberti <[EMAIL PROTECTED]> wrote:
> Is there a reason the variable ${DNIS} does not get set with incoming
> calls via chan_capi ?
>
> Is it related to the MSN=X in capi.conf ?

Just a guess, are you thinking of ${DNID} instead? There's no direct
mention of ${DNIS} on the wiki variables page, but ${DNID} works for
me with a BRI...

Andrew

--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett
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[Asterisk-Users] chan_capi setting ${DNIS}

2006-02-19 Thread Nathan Alberti




Is there a reason the variable ${DNIS} does not get set with incoming  
calls via chan_capi ?


Is it related to the MSN=X in capi.conf ?


version = chan_capi-cm-0.6.3

example;

exten => _9555XX,1,NoOp, ${EXTEN}, ${DNIS}



== ISDN1: Incoming call '04' -> '9555'
-- Executing SetCDRUserField("CAPI/ISDN1/95  55-135", "Incoming")  
in new stack

-- Executing NoOp("CAPI/ISDN1/9555-135", " 9555, ") in new stack
8< **SNIP**
> CAPI INFO 0x3490: Normal call clearing
== Spawn extension (main, s, 2) exited non-zero on 'CAPI/ 
ISDN1/9555-135'

== ISDN1: CAPI Hangingup
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RE: [Asterisk-Users] Chan_capi on builds 7955>8320 strangeness

2006-01-27 Thread gw

Strange though it's only effecting since build 8000...

Here's the snippet:

exten => s,1,LookupCIDName
exten =>
s,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/incoming/${IncomingLine
}/In-${STRFTIME(${EPOCH},,%Y%m% . . .
exten => s,3,Monitor(wav,${CALLFILENAME})
exten => s,4,GotoIf($["${INCOMINGLINE}" = "9146930821"]?9:5);
exten => s,5,GotoIfTime(20:01-7:59|mon-sun|*|*?9)
exten => s,6,Dial(${ADCOMDAYRINGTO},25,t);All
exten => s,7,NoOp(${DIALSTATUS})
exten => s,8,Goto(adcomincoming,s,11)
exten => s,9,Dial(${ADCOMNIGHTRINGTO},25,t);Cisco,Ping,Poly,SPA841
exten => s,10,NoOp(${DIALSTATUS})  
exten => s,107,Answer
exten => s,108,Wait(1)
exten => s,109,BackGround(adcom1/thankyou); Thank you for calling ADCOM
Corp.
exten => s,110,Playback(busy-pls-hold)
exten => s,111,Queue(adcomgwqueue)
exten => s,11,Answer ; Answer the line
exten => s,12,Wait(1)

... Menu plays.

ADCOMDAYRINGTO =
${C79601L1}&${OFFICE3}&${POLY1L1}&SIP/344&SIP/345&SIP/364&${SOMERSADCOM}
; SIP/355&SIP/342 SP
ADCOMNIGHTRINGTO =
${C79601L1}&${POLY1L1}&SIP/344&SIP/345&SIP/364&${SOMERSADCOM} 

So could it have something to do with the dialstring?  I would think
asterisk would say something first before doing the dials.

I'll try it later with a simple dialstring.  I'm going to rebuild it
anyhow.

I am looking to use a global variable in like a switch setup, to direct
calls to particular setups based on a menu.  For example, someone dials
ext 333, and they get a menu for day mode, night mode, holiday, away
from office, etc and the dialplan will ring different devices depending
on the choice...

We'll see what happens...

On a side note, I believe it works if I dial right into the menu
playback.  But if it's the dialstring that's wrong, I would think
asterisk should complain about it.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armin
Schindler
Sent: Friday, January 27, 2006 4:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Chan_capi on builds 7955>8320 strangeness

This is not a problem of the ISDN line (or chan_capi), Asterisk is just
not doing anything after

  -- Executing
GotoIfTime("CAPI/ISDNL1/5912211-0","20:01-7:59|mon-sun|*|*?9") in new
stack

and without further commands (like Ringing(), Answer(), ...) the ISDN
line timed out and disconnects.

So either your dialplan is buggy, or Asterisk is not doing what you
want.
What should be done according your extensions.conf in that state ?

Armin

On Fri, 27 Jan 2006 [EMAIL PROTECTED] wrote:
>  /etc/init.d/asterisk stop
> Stopping Asterisk PBX: .
> censys:/usr/src/asterisk-8632#  cd ..
> censys:/usr/src# asterisk -vc
> 
>   == Parsing '/etc/asterisk/asterisk.conf': Found
> 
>   == Parsing '/etc/asterisk/extconfig.conf': Found
> 
> Asterisk SVN-trunk-r8620, Copyright (C) 1999 - 2006 Digium, Inc. and 
> others.
> 
> Created by Mark Spencer <[EMAIL PROTECTED]>
> 
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for 
> details.
> 
> This is free software, with components licensed under the GNU General 
> Public
> 
> License version 2 and other licenses; you are welcome to redistribute 
> it under
> 
> certain conditions. Type 'show license' for details.
> 
> ==
> ==
> =
> 
>   == Parsing '/etc/asterisk/logger.conf': Found
> 
> Asterisk Event Logger Started /var/log/asterisk/event_log
> 
> Asterisk Dynamic Loader loading preload modules:
> 
> >>>>CLIP
>  [chan_capi.so] => (Common ISDN API for Asterisk)
> 
>   == Parsing '/etc/asterisk/capi.conf': Found
> 
>   == This box has 1 capi controller(s).
> 
> -- CAPI/contr1 supports DTMF
> 
> -- CAPI/contr1 supports echo cancellation
> 
> -- CAPI/contr1 supports line interconnect
> 
> -- CAPI/contr1 supports supplementary services
> 
>> supplementary services : 0x010f
> 
>> HOLD/RETRIEVE
> 
>> TERMINAL PORTABILITY
> 
>> ECT
> 
>> 3PTY
> 
>> MWI
> 
>   == Reading config for ISDNL1
> 
> -- capi_pvt ISDNL1-pseudo-D (5912211,capi-in-5912211,0,2) (1,4,64)
> 
> -- capi_pvt ISDNL1 (5912211,capi-in-5912211,0,2) (1,4,64)
> 
> -- capi_pvt ISDNL1 (5912211,capi-in-5912211,0,2) (1,4,64)
> 
>   == Reading config for ISDNL2
> 
> -- capi_pvt ISDNL2-pseudo-D (6930821,capi-in-6930821,0,2) (0,0,64)
> 
> -- capi_pvt ISDNL2 (6930821,capi-in-6930821,0,2) (0,0,64)
>

RE: [Asterisk-Users] Chan_capi on builds 7955>8320 strangeness

2006-01-27 Thread Armin Schindler

L1/5912211-0",
> "CALLFILENAME=/var/spool/asterisk/monitor/incoming/9145912211/In-2006011
> 8-030511-9145912211_ADCOM Ardlsey_s") in new stack
> 
>   == Started pbx on channel CAPI/ISDNL1/5912211-0
> 
>> CAPI devicestate requested for ISDNL1/5912211
> 
> -- Executing Monitor("CAPI/ISDNL1/5912211-0",
> "wav|/var/spool/asterisk/monitor/incoming/9145912211/In-20060118-030511-
> 9145912211_ADCOM Ardlsey_s") in new stack
> 
> -- Executing GotoIf("CAPI/ISDNL1/5912211-0", "0?9:5") in new stack
> 
> -- Goto (adcomincoming,s,5)
> 
> -- Executing GotoIfTime("CAPI/ISDNL1/5912211-0",
> "20:01-7:59|mon-sun|*|*?9") in new stack
> 
> -- IAX2/teliaxcsi-9 is ringing
> 
> INFO_IND ID=001 #0x0006 LEN=0015
>   Controller/PLCI/NCCI= 0x201
>   InfoNumber  = 0x804d
>   InfoElement = default
> 
> 
> INFO_RESP ID=001 #0x0006 LEN=0012
>   Controller/PLCI/NCCI= 0x201
> 
> 
> -- ISDNL1: info element RELEASE
> 
> DISCONNECT_IND ID=001 #0x0007 LEN=0014
>   Controller/PLCI/NCCI= 0x201
>   Reason  = 0x3490
> 
> 
> DISCONNECT_RESP ID=001 #0x0007 LEN=0012
>   Controller/PLCI/NCCI= 0x201
> 
> 
>> CAPI INFO 0x3490: Normal call clearing
> 
> -- Hungup 'IAX2/teliaxcsi-9'
> 
>   == Spawn extension (cisco-teliaxoutcsi, 19145912211, 4) exited
> non-zero on 'SIP/366-5e8d'
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Armin
> Schindler
> Sent: Friday, January 27, 2006 2:24 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Chan_capi on builds 7955>8320 strangeness
> 
> On Fri, 27 Jan 2006 [EMAIL PROTECTED] wrote:
> > Hello All,
> > I am having an odd problem with Armin's chan-capi_cm on builds higher 
> > than 7955.
> > 
> > It would seem that this happens on anything higher than 7955.
> > 
> > What is happening is the isdn is ringing, then asterisk does a goto-if
> 
> > and just hangs.
> > 
> > Asterisk itself is ok, but the isdn then rings out or busys out on the
> 
> > other side.
> > 
> > Outgoing works fine, this only seems to effect incoming.
> > 
> > I updated to chan-capi_cm 0.6.3 but there is no change.
> > 
> > Noticed this when trying to update for the timebomb bug.
> > 
> > I think it is somehow related to the dial command but I'm not certain.
> > 
> > Has anyone else experienced such oddness?
> 
> Can you please create a log (set verbose 5, capi debug)?
> 
> Armin
> 
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RE: [Asterisk-Users] Chan_capi on builds 7955>8320 strangeness

2006-01-27 Thread gw

2211
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BC  = <80 90 a2>
  LLC = default
  HLC = default
  AdditionalInfo 
   BChannelinformation= default
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default


-- CONNECT_IND
(PLCI=0x201,DID=5912211,CID=9145912211,CIP=0x1,CONTROLLER=0x1)

   > ISDNL2: msn='6930821' DNID='5912211' MSN

   > ISDNL2: msn='6930821' DNID='5912211' MSN

   > ISDNL1: msn='5912211' DNID='5912211' MSN

  == ISDNL1: Incoming call '19145912211' -> '5912211'

INFO_IND ID=001 #0x0002 LEN=0023
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x70
  InfoElement = 5912211


INFO_RESP ID=001 #0x0002 LEN=0012
  Controller/PLCI/NCCI= 0x201


-- ISDNL1: info element CALLED PARTY NUMBER

   > ISDNL1: INFO_IND DID digits not used in this state.

INFO_IND ID=001 #0x0003 LEN=0025
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x28
  InfoElement = 9145912211


INFO_RESP ID=001 #0x0003 LEN=0012
  Controller/PLCI/NCCI= 0x201


-- ISDNL1: info element DSP

INFO_IND ID=001 #0x0004 LEN=0016
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x18
  InfoElement = <89>


INFO_RESP ID=001 #0x0004 LEN=0012
  Controller/PLCI/NCCI= 0x201


-- ISDNL1: info element CHANNEL IDENTIFICATION 89

INFO_IND ID=001 #0x0005 LEN=0015
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x8005
  InfoElement = default


INFO_RESP ID=001 #0x0005 LEN=0012
  Controller/PLCI/NCCI= 0x201


-- ISDNL1: info element SETUP

-- ISDNL1: CAPI/ISDNL1/5912211-0: 5912211 matches in context
capi-in-5912211

-- Executing Set("CAPI/ISDNL1/5912211-0", "IncomingCID=""
<19145912211>") in new stack

-- Executing Set("CAPI/ISDNL1/5912211-0", "IncomingLine=9145912211")
in new stack

-- Executing Set("CAPI/ISDNL1/5912211-0", "CALLERID(name)=IN 2211")
in new stack

-- Executing GotoIf("CAPI/ISDNL1/5912211-0", "1?5:6") in new stack

-- Goto (capi-in-5912211,5912211,5)

-- Executing Set("CAPI/ISDNL1/5912211-0",
"CALLERID(num)=9145912211") in new stack

-- Executing GotoIf("CAPI/ISDNL1/5912211-0", "1?16") in new stack

-- Goto (capi-in-5912211,5912211,16)

-- Executing Goto("CAPI/ISDNL1/5912211-0", "adcomincoming|s|1") in
new stack

-- Goto (adcomincoming,s,1)

-- Executing LookupCIDName("CAPI/ISDNL1/5912211-0", "") in new stack

-- Changed Caller*ID name to ADCOM Ardlsey

-- Executing Set("CAPI/ISDNL1/5912211-0",
"CALLFILENAME=/var/spool/asterisk/monitor/incoming/9145912211/In-2006011
8-030511-9145912211_ADCOM Ardlsey_s") in new stack

  == Started pbx on channel CAPI/ISDNL1/5912211-0

   > CAPI devicestate requested for ISDNL1/5912211

-- Executing Monitor("CAPI/ISDNL1/5912211-0",
"wav|/var/spool/asterisk/monitor/incoming/9145912211/In-20060118-030511-
9145912211_ADCOM Ardlsey_s") in new stack

-- Executing GotoIf("CAPI/ISDNL1/5912211-0", "0?9:5") in new stack

-- Goto (adcomincoming,s,5)

-- Executing GotoIfTime("CAPI/ISDNL1/5912211-0",
"20:01-7:59|mon-sun|*|*?9") in new stack

-- IAX2/teliaxcsi-9 is ringing

INFO_IND ID=001 #0x0006 LEN=0015
  Controller/PLCI/NCCI= 0x201
  InfoNumber  = 0x804d
  InfoElement         = default


INFO_RESP ID=001 #0x0006 LEN=0012
  Controller/PLCI/NCCI= 0x201


-- ISDNL1: info element RELEASE

DISCONNECT_IND ID=001 #0x0007 LEN=0014
  Controller/PLCI/NCCI= 0x201
  Reason  = 0x3490


DISCONNECT_RESP ID=001 #0x0007 LEN=0012
  Controller/PLCI/NCCI= 0x201


   > CAPI INFO 0x3490: Normal call clearing

-- Hungup 'IAX2/teliaxcsi-9'

  == Spawn extension (cisco-teliaxoutcsi, 19145912211, 4) exited
non-zero on 'SIP/366-5e8d'

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armin
Schindler
Sent: Friday, January 27, 2006 2:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Chan_capi on builds 7955>8320 strangeness

On Fri, 27 Jan 2006 [EMAIL PROTECTED] wrote:
> Hello All,
> I am having an odd problem with Armin's chan-capi_cm on builds higher 
> than 7955.
> 
&

Re: [Asterisk-Users] Chan_capi on builds 7955>8320 strangeness

2006-01-26 Thread Armin Schindler

On Fri, 27 Jan 2006 [EMAIL PROTECTED] wrote:
> Hello All,
> I am having an odd problem with Armin's chan-capi_cm on builds higher
> than 7955.
> 
> It would seem that this happens on anything higher than 7955.
> 
> What is happening is the isdn is ringing, then asterisk does a goto-if
> and just hangs.  
> 
> Asterisk itself is ok, but the isdn then rings out or busys out on the
> other side.
> 
> Outgoing works fine, this only seems to effect incoming.
> 
> I updated to chan-capi_cm 0.6.3 but there is no change.
> 
> Noticed this when trying to update for the timebomb bug.
> 
> I think it is somehow related to the dial command but I'm not certain.
> 
> Has anyone else experienced such oddness?

Can you please create a log (set verbose 5, capi debug)?

Armin

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[Asterisk-Users] Chan_capi on builds 7955>8320 strangeness

2006-01-26 Thread gw

Hello All,
I am having an odd problem with Armin's chan-capi_cm on builds higher
than 7955.

It would seem that this happens on anything higher than 7955.

What is happening is the isdn is ringing, then asterisk does a goto-if
and just hangs.  

Asterisk itself is ok, but the isdn then rings out or busys out on the
other side.

Outgoing works fine, this only seems to effect incoming.

I updated to chan-capi_cm 0.6.3 but there is no change.

Noticed this when trying to update for the timebomb bug.

I think it is somehow related to the dial command but I'm not certain.

Has anyone else experienced such oddness?

Regards,
Greg
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Re: [Asterisk-Users] chan_capi - B3 Error

2006-01-24 Thread Nathan Alberti



On 24/01/2006, at 5:55 PM, Armin Schindler wrote:


On Tue, 24 Jan 2006, Nathan Alberti wrote:

Thank you Armin,

Yes, it is a fritz card :)

I till try with the overlap settings later today.

Let me know if I can be of any assistance with debug info or  
anything you

need.


Thanks for the offer, I will come back to that. Maybe you can test the
following patch when using dial option 'b' only:

--- chan_capi.c 6 Jan 2006 14:35:51 -   1.175
+++ chan_capi.c 9 Jan 2006 20:17:00 -
@@ -2454,7 +2454,6 @@
case 0x8002:/* CALL PROCEEDING */
 		cc_verbose(3, 1, VERBOSE_PREFIX_3 "%s: info element CALL  
PROCEEDING\n",

i->name);
-   start_early_b3(i);
fr.frametype = AST_FRAME_CONTROL;
fr.subclass = AST_CONTROL_PROCEEDING;
pipe_frame(i, &fr);


Armin



Looks good.. better than my keyboard which got covered in breakfast  
cereal while I was working on it :)



-- Called g1/142392203034/b
-- CAPI/ISDN1/142392203034-4 is proceeding passing it to SIP/ 
0014A8ACCB83-790a

-- CAPI/ISDN1/142392203034-4 is ringing
-- CAPI/ISDN1/142392203034-4 is making progress passing it to  
SIP/0014A8ACCB83-790a

-- CAPI/ISDN1/142392203034-4 answered SIP/0014A8ACCB83-790a

Call progress tones are being passed back successfully and no more  
errors.


Regards,

Nathan

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Re: [Asterisk-Users] chan_capi - B3 Error

2006-01-24 Thread Armin Schindler

On Tue, 24 Jan 2006, Nathan Alberti wrote:
> Thank you Armin,
> 
> Yes, it is a fritz card :)
> 
> I till try with the overlap settings later today.
> 
> Let me know if I can be of any assistance with debug info or anything you
> need.

Thanks for the offer, I will come back to that. Maybe you can test the 
following patch when using dial option 'b' only:

--- chan_capi.c 6 Jan 2006 14:35:51 -   1.175
+++ chan_capi.c 9 Jan 2006 20:17:00 -
@@ -2454,7 +2454,6 @@
case 0x8002:/* CALL PROCEEDING */
cc_verbose(3, 1, VERBOSE_PREFIX_3 "%s: info element CALL 
PROCEEDING\n",
i->name);
-   start_early_b3(i);
fr.frametype = AST_FRAME_CONTROL;
fr.subclass = AST_CONTROL_PROCEEDING;
pipe_frame(i, &fr);


Armin

> On 24/01/2006, at 3:42 PM, Armin Schindler wrote:
> 
> > Let e guess, you have an AVM card?
> > 
> > It is a known issue. In some cases the driver does no accept the B
> > connect
> > request, which would be okay, but any try later when it must be possible,
> > it
> > is rejected too.
> > I have this on my todo list, but in the meantime it should work fine when
> > you switch to overlap dial using options 'bo' instead of only 'b' in the
> > CAPI dial string.
> > 
> > Armin
> > 
> > On Tue, 24 Jan 2006, Nathan Alberti wrote:
> > > I seem to be having a problem with B3 on my ISDN line, as you can see
> > > from the
> > > dial string I am having to have asterisk generate ringing else there
> > > is no
> > > progress indication.
> > > 
> > > 
> > >-- Executing Dial("SIP/0014A8ACCB83-fd9f", "CAPI/
> > > g1/142392203000/b|40|r")
> > > in new stack
> > > -- Called g1/142392203000/b
> > > -- CAPI/ISDN1/142392203000-0 is proceeding passing it to SIP/
> > > 0014A8ACCB83-fd9f
> > > Jan 24 07:38:56 WARNING[10609]: chan_capi.c:3385
> > > show_capi_conf_error: ISDN1:
> > > conf_error 0x2001 PLCI=0x101 Command=CONNECT_B3_CONF,0x8487
> > > -- CAPI/ISDN1/142392203000-0 answered SIP/0014A8ACCB83-fd9f
> > > 
> > > This issue was only introduced after and upgrade to chan_capi-
> > > cm-0.6.1 and
> > > continues on to chan_capi-cm-0.6.3, my capi.conf is as follows;
> > > 
> > > 
> > > [general]
> > > nationalprefix=0
> > > internationalprefix=0
> > > rxgain=0.8
> > > txgain=0.8
> > > 
> > > [ISDN1]
> > > isdnmode=msn
> > > controller=1
> > > group=1
> > > softdtmf=0
> > > relaxdtmf=on
> > > context=pstn_in
> > > callgroup=1
> > > devices=2
> > > 
> > > 
> > > Regards,
> > > 
> > > Nathan.
> > > ___
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Re: [Asterisk-Users] chan_capi - B3 Error

2006-01-24 Thread Nathan Alberti


Thank you Armin,

Yes, it is a fritz card :)

I till try with the overlap settings later today.

Let me know if I can be of any assistance with debug info or anything  
you need.


Regards,

Nathan.

On 24/01/2006, at 3:42 PM, Armin Schindler wrote:


Let e guess, you have an AVM card?

It is a known issue. In some cases the driver does no accept the B  
connect
request, which would be okay, but any try later when it must be  
possible, it

is rejected too.
I have this on my todo list, but in the meantime it should work  
fine when
you switch to overlap dial using options 'bo' instead of only 'b'  
in the

CAPI dial string.

Armin

On Tue, 24 Jan 2006, Nathan Alberti wrote:
I seem to be having a problem with B3 on my ISDN line, as you can  
see from the
dial string I am having to have asterisk generate ringing else  
there is no

progress indication.


-- Executing Dial("SIP/0014A8ACCB83-fd9f", "CAPI/ 
g1/142392203000/b|40|r")

in new stack
-- Called g1/142392203000/b
-- CAPI/ISDN1/142392203000-0 is proceeding passing it to SIP/
0014A8ACCB83-fd9f
Jan 24 07:38:56 WARNING[10609]: chan_capi.c:3385  
show_capi_conf_error: ISDN1:

conf_error 0x2001 PLCI=0x101 Command=CONNECT_B3_CONF,0x8487
-- CAPI/ISDN1/142392203000-0 answered SIP/0014A8ACCB83-fd9f

This issue was only introduced after and upgrade to chan_capi- 
cm-0.6.1 and

continues on to chan_capi-cm-0.6.3, my capi.conf is as follows;


[general]
nationalprefix=0
internationalprefix=0
rxgain=0.8
txgain=0.8

[ISDN1]
isdnmode=msn
controller=1
group=1
softdtmf=0
relaxdtmf=on
context=pstn_in
callgroup=1
devices=2


Regards,

Nathan.
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Re: [Asterisk-Users] chan_capi - B3 Error

2006-01-23 Thread Armin Schindler

Let e guess, you have an AVM card?

It is a known issue. In some cases the driver does no accept the B connect 
request, which would be okay, but any try later when it must be possible, it
is rejected too.
I have this on my todo list, but in the meantime it should work fine when 
you switch to overlap dial using options 'bo' instead of only 'b' in the
CAPI dial string.

Armin

On Tue, 24 Jan 2006, Nathan Alberti wrote:
> I seem to be having a problem with B3 on my ISDN line, as you can see from the
> dial string I am having to have asterisk generate ringing else there is no
> progress indication.
> 
> 
> -- Executing Dial("SIP/0014A8ACCB83-fd9f", "CAPI/g1/142392203000/b|40|r")
> in new stack
> -- Called g1/142392203000/b
> -- CAPI/ISDN1/142392203000-0 is proceeding passing it to SIP/
> 0014A8ACCB83-fd9f
> Jan 24 07:38:56 WARNING[10609]: chan_capi.c:3385 show_capi_conf_error: ISDN1:
> conf_error 0x2001 PLCI=0x101 Command=CONNECT_B3_CONF,0x8487
> -- CAPI/ISDN1/142392203000-0 answered SIP/0014A8ACCB83-fd9f
> 
> This issue was only introduced after and upgrade to chan_capi-cm-0.6.1 and
> continues on to chan_capi-cm-0.6.3, my capi.conf is as follows;
> 
> 
> [general]
> nationalprefix=0
> internationalprefix=0
> rxgain=0.8
> txgain=0.8
> 
> [ISDN1]
> isdnmode=msn
> controller=1
> group=1
> softdtmf=0
> relaxdtmf=on
> context=pstn_in
> callgroup=1
> devices=2
> 
> 
> Regards,
> 
> Nathan.
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[Asterisk-Users] chan_capi - B3 Error

2006-01-23 Thread Nathan Alberti



I seem to be having a problem with B3 on my ISDN line, as you can see  
from the dial string I am having to have asterisk generate ringing  
else there is no progress indication.



-- Executing Dial("SIP/0014A8ACCB83-fd9f", "CAPI/g1/142392203000/ 
b|40|r") in new stack

-- Called g1/142392203000/b
-- CAPI/ISDN1/142392203000-0 is proceeding passing it to SIP/ 
0014A8ACCB83-fd9f
Jan 24 07:38:56 WARNING[10609]: chan_capi.c:3385  
show_capi_conf_error: ISDN1: conf_error 0x2001 PLCI=0x101  
Command=CONNECT_B3_CONF,0x8487

-- CAPI/ISDN1/142392203000-0 answered SIP/0014A8ACCB83-fd9f

This issue was only introduced after and upgrade to chan_capi- 
cm-0.6.1 and continues on to chan_capi-cm-0.6.3, my capi.conf is as  
follows;



[general]
nationalprefix=0
internationalprefix=0
rxgain=0.8
txgain=0.8

[ISDN1]
isdnmode=msn
controller=1
group=1
softdtmf=0
relaxdtmf=on
context=pstn_in
callgroup=1
devices=2


Regards,

Nathan.
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Re: [Asterisk-Users] chan_capi-cm and DID

2006-01-19 Thread richard Coco


Hi armin,

thx for the answer. I have connected the BRI on a
HiPazt4000 and i still have the same issue. So i think
i have a problem with my ISDN line. I will contact my
provider. May be a reset of the line will solve the
problem.

rich.

--- Armin Schindler <[EMAIL PROTECTED]> wrote:

> On Tue, 17 Jan 2006, richard Coco wrote:
> > Hi Armin,
> > 
> > thx for your feedback, but what do you mean with
> "Did
> > you load the card with config for DID on that
> port?"
> > 
> > I have loaded the modules with:
> > modprobe capi 
> > modprobe kernelcapi 
> > modprobe divacapi 
> > modprobe divas
> > 
> > and then loaded divactrl like this:
> > divactrl load -f ETSI
> > 
> > I suppose that this is ok (it works without did)?
> Or
> > have i forgotten something?
> 
> With 
>   divactrl load -f ETSI
> you load the card to PtMP (which is the default) on
> all four ports.
> Use
>   divactrl load -c 1 -SeparateConfig -u1
> where the '1' of -u1 means second port.
> E.g. -u is first port, -u1 -u2 -u3 is port 2,3,4.
> 
> When using -SeparateConfig, the X-extension is
> available
> for many options.
> 
> E.g., you can put port 3 and 4 into NT-mode, or even
> run another protocol
> (1TR6, JAPAN, QSIG,...) on other ports.
> 
> See
>   divactrl load -h
> for all options.
>  
> Armin
> 
> > thx in advance..
> > 
> > --- Armin Schindler <[EMAIL PROTECTED]> wrote:
> > 
> > > On Mon, 16 Jan 2006, richard Coco wrote:
> > > > Hi all,
> > > > 
> > > > i have asterisk 1.0.9 with an Eicon Diva 4bri
> and
> > > > chan_capi-cm-0.6. I have 2 NTBAs (one with did
> and
> > > one
> > > > without).
> > > > When using the one without did, i am able to
> place
> > > > outgoing and incoming calls. When i use the
> NTBAs
> > > with
> > > > did i have a layer 2 error.
> > > > 
> > > > Anyone an idea?
> > > 
> > > Did you load the card with config for DID on
> that
> > > port?
> > > What are your divactrl parameters? (Or do you
> use
> > > Eicon Package with xml based config?)
> > > 
> > > Armin 
> > >  
> > > > -- Executing Dial("SIP/2004-9634",
> > > > "CAPI/g1/43XX") in new stack
> > > >> data = g1/43XX
> > > >> parsed dialstring: 'g1' '43XX' ''
> > > >> capi request group = 2
> > > >> parsed dialstring: 'g1' '43XX' ''
> > > >   == EICON: Call CAPI/EICON/43XX-6  
> > > (pres=0x00,
> > > > ton=0x00)
> > > > CONNECT_REQ ID=001 #0x000c LEN=0065
> > > >   Controller/PLCI/NCCI= 0x1
> > > >   CIPValue= 0x10
> > > >   CalledPartyNumber   =
> <80>43XX
> > > >   CallingPartyNumber  = <00 80
> > > > 22>EyeBeam<22> <3c>2004<3e>
> > > >   CalledPartySubaddress   = default
> > > >   CallingPartySubaddress  = default
> > > >   BProtocol
> > > >B1protocol = 0x1
> > > >B2protocol = 0x1
> > > >B3protocol = 0x0
> > > >B1configuration= default
> > > >B2configuration= default
> > > >B3configuration= default
> > > >   BC  = default
> > > >   LLC = default
> > > >   HLC = default
> > > >   AdditionalInfo
> > > >BChannelinformation= <00 00>
> > > >Keypadfacility = default
> > > >Useruserdata   = default
> > > >Facilitydataarray  = default
> > > > 
> > > > -- Called g1/43XX
> > > > CONNECT_CONF ID=001 #0x000c LEN=0014
> > > >   Controller/PLCI/NCCI= 0x201
> > > >   Info= 0x0
> > > > 
> > > > -- EICON: received CONNECT_CONF PLCI =
> 0x201
> > > > DISCONNECT_IND ID=001 #0x0011 LEN=0014
> > > >   Controller/PLCI/NCCI= 0x201
> > > >   Reason  = 0x3302
> > > > 
> > > > DISCONNECT_RESP ID=001 #0x0011 LEN=0012
> > > >   Controller/PLCI/NCCI= 0x201
> > > > 
> > > >> CAPI INFO 0x3302: Protocol error
> layer 2
> > > >   == EICON: CAPI Hangingup
> > > >   == EICON: Interface cleanup PLCI=0x201
> > > >   == No one is available to answer at this
> time
> > > > 
> > > > my capi.conf looks like:
> > > > [DID]
> > > > controller=1,2,3,4
> > > > isdnmode=did
> > > > incomingmsn=*
> > > > softdtmf=on
> > > > relaxdtmf=on
> > > > accountcode=
> > > > context=DID
> > > > echocancel=yes
> > > > ;echocancelold=yes
> > > > devices=2
> > > > group=1
> > > > 
> > > > 
> > > > 
> > > >
> __
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> > > protection around 
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> > > >
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> > > >   
> > >
> >
>
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> > > > 
>

Re: [Asterisk-Users] chan_capi-cm and DID

2006-01-17 Thread Armin Schindler

On Tue, 17 Jan 2006, richard Coco wrote:
> Hi Armin,
> 
> thx for your feedback, but what do you mean with "Did
> you load the card with config for DID on that port?"
> 
> I have loaded the modules with:
> modprobe capi 
> modprobe kernelcapi 
> modprobe divacapi 
> modprobe divas
> 
> and then loaded divactrl like this:
> divactrl load -f ETSI
> 
> I suppose that this is ok (it works without did)? Or
> have i forgotten something?

With 
  divactrl load -f ETSI
you load the card to PtMP (which is the default) on all four ports.
Use
  divactrl load -c 1 -SeparateConfig -u1
where the '1' of -u1 means second port.
E.g. -u is first port, -u1 -u2 -u3 is port 2,3,4.

When using -SeparateConfig, the X-extension is available
for many options.

E.g., you can put port 3 and 4 into NT-mode, or even run another protocol
(1TR6, JAPAN, QSIG,...) on other ports.

See
  divactrl load -h
for all options.
 
Armin

> thx in advance..
> 
> --- Armin Schindler <[EMAIL PROTECTED]> wrote:
> 
> > On Mon, 16 Jan 2006, richard Coco wrote:
> > > Hi all,
> > > 
> > > i have asterisk 1.0.9 with an Eicon Diva 4bri and
> > > chan_capi-cm-0.6. I have 2 NTBAs (one with did and
> > one
> > > without).
> > > When using the one without did, i am able to place
> > > outgoing and incoming calls. When i use the NTBAs
> > with
> > > did i have a layer 2 error.
> > > 
> > > Anyone an idea?
> > 
> > Did you load the card with config for DID on that
> > port?
> > What are your divactrl parameters? (Or do you use
> > Eicon Package with xml based config?)
> > 
> > Armin 
> >  
> > > -- Executing Dial("SIP/2004-9634",
> > > "CAPI/g1/43XX") in new stack
> > >> data = g1/43XX
> > >> parsed dialstring: 'g1' '43XX' ''
> > >> capi request group = 2
> > >> parsed dialstring: 'g1' '43XX' ''
> > >   == EICON: Call CAPI/EICON/43XX-6  
> > (pres=0x00,
> > > ton=0x00)
> > > CONNECT_REQ ID=001 #0x000c LEN=0065
> > >   Controller/PLCI/NCCI= 0x1
> > >   CIPValue= 0x10
> > >   CalledPartyNumber   = <80>43XX
> > >   CallingPartyNumber  = <00 80
> > > 22>EyeBeam<22> <3c>2004<3e>
> > >   CalledPartySubaddress   = default
> > >   CallingPartySubaddress  = default
> > >   BProtocol
> > >B1protocol = 0x1
> > >B2protocol = 0x1
> > >B3protocol = 0x0
> > >B1configuration= default
> > >B2configuration= default
> > >B3configuration= default
> > >   BC  = default
> > >   LLC = default
> > >   HLC = default
> > >   AdditionalInfo
> > >BChannelinformation= <00 00>
> > >Keypadfacility = default
> > >Useruserdata   = default
> > >Facilitydataarray  = default
> > > 
> > > -- Called g1/43XX
> > > CONNECT_CONF ID=001 #0x000c LEN=0014
> > >   Controller/PLCI/NCCI= 0x201
> > >   Info= 0x0
> > > 
> > > -- EICON: received CONNECT_CONF PLCI = 0x201
> > > DISCONNECT_IND ID=001 #0x0011 LEN=0014
> > >   Controller/PLCI/NCCI= 0x201
> > >   Reason  = 0x3302
> > > 
> > > DISCONNECT_RESP ID=001 #0x0011 LEN=0012
> > >   Controller/PLCI/NCCI= 0x201
> > > 
> > >> CAPI INFO 0x3302: Protocol error layer 2
> > >   == EICON: CAPI Hangingup
> > >   == EICON: Interface cleanup PLCI=0x201
> > >   == No one is available to answer at this time
> > > 
> > > my capi.conf looks like:
> > > [DID]
> > > controller=1,2,3,4
> > > isdnmode=did
> > > incomingmsn=*
> > > softdtmf=on
> > > relaxdtmf=on
> > > accountcode=
> > > context=DID
> > > echocancel=yes
> > > ;echocancelold=yes
> > > devices=2
> > > group=1
> > > 
> > > 
> > > 
> > > __
> > > Do You Yahoo!?
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> > protection around 
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> >
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Re: [Asterisk-Users] chan_capi-cm and DID

2006-01-17 Thread richard Coco

Hi Armin,

thx for your feedback, but what do you mean with "Did
you load the card with config for DID on that port?"

I have loaded the modules with:
modprobe capi 
modprobe kernelcapi 
modprobe divacapi 
modprobe divas

and then loaded divactrl like this:
divactrl load -f ETSI

I suppose that this is ok (it works without did)? Or
have i forgotten something?

thx in advance..

--- Armin Schindler <[EMAIL PROTECTED]> wrote:

> On Mon, 16 Jan 2006, richard Coco wrote:
> > Hi all,
> > 
> > i have asterisk 1.0.9 with an Eicon Diva 4bri and
> > chan_capi-cm-0.6. I have 2 NTBAs (one with did and
> one
> > without).
> > When using the one without did, i am able to place
> > outgoing and incoming calls. When i use the NTBAs
> with
> > did i have a layer 2 error.
> > 
> > Anyone an idea?
> 
> Did you load the card with config for DID on that
> port?
> What are your divactrl parameters? (Or do you use
> Eicon Package with xml based config?)
> 
> Armin 
>  
> > -- Executing Dial("SIP/2004-9634",
> > "CAPI/g1/43XX") in new stack
> >> data = g1/43XX
> >> parsed dialstring: 'g1' '43XX' ''
> >> capi request group = 2
> >> parsed dialstring: 'g1' '43XX' ''
> >   == EICON: Call CAPI/EICON/43XX-6  
> (pres=0x00,
> > ton=0x00)
> > CONNECT_REQ ID=001 #0x000c LEN=0065
> >   Controller/PLCI/NCCI= 0x1
> >   CIPValue= 0x10
> >   CalledPartyNumber   = <80>43XX
> >   CallingPartyNumber  = <00 80
> > 22>EyeBeam<22> <3c>2004<3e>
> >   CalledPartySubaddress   = default
> >   CallingPartySubaddress  = default
> >   BProtocol
> >B1protocol = 0x1
> >B2protocol = 0x1
> >B3protocol = 0x0
> >B1configuration= default
> >B2configuration= default
> >B3configuration= default
> >   BC  = default
> >   LLC = default
> >   HLC = default
> >   AdditionalInfo
> >BChannelinformation= <00 00>
> >Keypadfacility = default
> >Useruserdata   = default
> >Facilitydataarray  = default
> > 
> > -- Called g1/43XX
> > CONNECT_CONF ID=001 #0x000c LEN=0014
> >   Controller/PLCI/NCCI= 0x201
> >   Info= 0x0
> > 
> > -- EICON: received CONNECT_CONF PLCI = 0x201
> > DISCONNECT_IND ID=001 #0x0011 LEN=0014
> >   Controller/PLCI/NCCI= 0x201
> >   Reason  = 0x3302
> > 
> > DISCONNECT_RESP ID=001 #0x0011 LEN=0012
> >   Controller/PLCI/NCCI= 0x201
> > 
> >> CAPI INFO 0x3302: Protocol error layer 2
> >   == EICON: CAPI Hangingup
> >   == EICON: Interface cleanup PLCI=0x201
> >   == No one is available to answer at this time
> > 
> > my capi.conf looks like:
> > [DID]
> > controller=1,2,3,4
> > isdnmode=did
> > incomingmsn=*
> > softdtmf=on
> > relaxdtmf=on
> > accountcode=
> > context=DID
> > echocancel=yes
> > ;echocancelold=yes
> > devices=2
> > group=1
> > 
> > 
> > 
> > __
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> > http://mail.yahoo.com 
> > ___
> > --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
> > 
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> --
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> To UNSUBSCRIBE or update options visit:
>   
>
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Re: [Asterisk-Users] chan_capi-cm and DID

2006-01-16 Thread Armin Schindler

On Mon, 16 Jan 2006, richard Coco wrote:
> Hi all,
> 
> i have asterisk 1.0.9 with an Eicon Diva 4bri and
> chan_capi-cm-0.6. I have 2 NTBAs (one with did and one
> without).
> When using the one without did, i am able to place
> outgoing and incoming calls. When i use the NTBAs with
> did i have a layer 2 error.
> 
> Anyone an idea?

Did you load the card with config for DID on that port?
What are your divactrl parameters? (Or do you use
Eicon Package with xml based config?)

Armin 
 
> -- Executing Dial("SIP/2004-9634",
> "CAPI/g1/43XX") in new stack
>> data = g1/43XX
>> parsed dialstring: 'g1' '43XX' ''
>> capi request group = 2
>> parsed dialstring: 'g1' '43XX' ''
>   == EICON: Call CAPI/EICON/43XX-6   (pres=0x00,
> ton=0x00)
> CONNECT_REQ ID=001 #0x000c LEN=0065
>   Controller/PLCI/NCCI= 0x1
>   CIPValue= 0x10
>   CalledPartyNumber   = <80>43XX
>   CallingPartyNumber  = <00 80
> 22>EyeBeam<22> <3c>2004<3e>
>   CalledPartySubaddress   = default
>   CallingPartySubaddress  = default
>   BProtocol
>B1protocol = 0x1
>B2protocol = 0x1
>B3protocol = 0x0
>B1configuration= default
>B2configuration= default
>B3configuration= default
>   BC  = default
>   LLC = default
>   HLC = default
>   AdditionalInfo
>BChannelinformation= <00 00>
>Keypadfacility = default
>Useruserdata   = default
>Facilitydataarray  = default
> 
> -- Called g1/43XX
> CONNECT_CONF ID=001 #0x000c LEN=0014
>   Controller/PLCI/NCCI= 0x201
>   Info= 0x0
> 
> -- EICON: received CONNECT_CONF PLCI = 0x201
> DISCONNECT_IND ID=001 #0x0011 LEN=0014
>   Controller/PLCI/NCCI= 0x201
>   Reason  = 0x3302
> 
> DISCONNECT_RESP ID=001 #0x0011 LEN=0012
>   Controller/PLCI/NCCI= 0x201
> 
>> CAPI INFO 0x3302: Protocol error layer 2
>   == EICON: CAPI Hangingup
>   == EICON: Interface cleanup PLCI=0x201
>   == No one is available to answer at this time
> 
> my capi.conf looks like:
> [DID]
> controller=1,2,3,4
> isdnmode=did
> incomingmsn=*
> softdtmf=on
> relaxdtmf=on
> accountcode=
> context=DID
> echocancel=yes
> ;echocancelold=yes
> devices=2
> group=1
> 
> 
> 
> __
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around 
> http://mail.yahoo.com 
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> --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
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[Asterisk-Users] chan_capi-cm and DID

2006-01-16 Thread richard Coco

Hi all,

i have asterisk 1.0.9 with an Eicon Diva 4bri and
chan_capi-cm-0.6. I have 2 NTBAs (one with did and one
without).
When using the one without did, i am able to place
outgoing and incoming calls. When i use the NTBAs with
did i have a layer 2 error.

Anyone an idea?


-- Executing Dial("SIP/2004-9634",
"CAPI/g1/43XX") in new stack
   > data = g1/43XX
   > parsed dialstring: 'g1' '43XX' ''
   > capi request group = 2
   > parsed dialstring: 'g1' '43XX' ''
  == EICON: Call CAPI/EICON/43XX-6   (pres=0x00,
ton=0x00)
CONNECT_REQ ID=001 #0x000c LEN=0065
  Controller/PLCI/NCCI= 0x1
  CIPValue= 0x10
  CalledPartyNumber   = <80>43XX
  CallingPartyNumber  = <00 80
22>EyeBeam<22> <3c>2004<3e>
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BProtocol
   B1protocol = 0x1
   B2protocol = 0x1
   B3protocol = 0x0
   B1configuration= default
   B2configuration= default
   B3configuration= default
  BC  = default
  LLC = default
  HLC = default
  AdditionalInfo
   BChannelinformation= <00 00>
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default

-- Called g1/43XX
CONNECT_CONF ID=001 #0x000c LEN=0014
  Controller/PLCI/NCCI= 0x201
  Info= 0x0

-- EICON: received CONNECT_CONF PLCI = 0x201
DISCONNECT_IND ID=001 #0x0011 LEN=0014
  Controller/PLCI/NCCI= 0x201
  Reason  = 0x3302

DISCONNECT_RESP ID=001 #0x0011 LEN=0012
  Controller/PLCI/NCCI= 0x201

   > CAPI INFO 0x3302: Protocol error layer 2
  == EICON: CAPI Hangingup
  == EICON: Interface cleanup PLCI=0x201
  == No one is available to answer at this time

my capi.conf looks like:
[DID]
controller=1,2,3,4
isdnmode=did
incomingmsn=*
softdtmf=on
relaxdtmf=on
accountcode=
context=DID
echocancel=yes
;echocancelold=yes
devices=2
group=1



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Re: [Asterisk-Users] CHAN_CAPI problem

2006-01-14 Thread Armin Schindler

On Wed, 11 Jan 2006 [EMAIL PROTECTED] wrote:
> 
> 
> Ok it solved my problem  (immediate=yes in capi.conf) !!!
> 
> Here is the console log
> ***
> CONNECT_IND ID=002 #0x201f LEN=0047
...
> ***
> 
> What is the meaning of immediate=yes ? I read the comment , but I was not
> able to understand it;
> In other words: in a ;'MSN' (point-to-multipoint)  (not DID) configuration,
> it is always required "immediate=yes" or is it due to a particular
> configuration of my Telecom ISDN BRI ? If it is always required, I could
> suggest you to put a comment about this on the capi.conf..

It is not always necessary. I cannot tell for sure, but it seems some 
drivers don't produce the needed indication for SETUP/SENDING-COMPLETE.
And if that's the case you need immediate=yes not to wait for these 
indications. Other drivers (like Eicon DIVA) do signal these indications
after all other indications of the incoming-call/setup message like 
REDIRECTING-NUMBER. So if you have such a faulty driver, the indications for
e.g. REDIRECTING-NUMBER may come after the call is processed (given to 
Asterisk) and the variable REDIRECTINGNUMBER is not set correctly.

Armin

> 
>
>  Armin Schindler   
>  <[EMAIL PROTECTED] 
>  >  To 
>  Sent by:  Asterisk Users Mailing List -   
>  asterisk-users-bo Non-Commercial Discussion   
>  [EMAIL PROTECTED]
>  m.com  cc 
>
>        Subject 
>  11/01/2006 10.24  Re: [Asterisk-Users] CHAN_CAPI  
>problem 
>
>  Please respond to 
>   Asterisk Users   
>   Mailing List -   
>   Non-Commercial   
> Discussion 
>  <[EMAIL PROTECTED] 
>  ists.digium.com>  
>
>
> 
> 
> 
> 
> There is no 'sending-complete'/'setup' info-element, please use
> immediate=yes in capi.conf
> 
> Armin
> 
> 
> On Wed, 11 Jan 2006 [EMAIL PROTECTED] wrote:
> > Thank you very much for your attention;
> > Here is what you asked for:
> >
> ***
> 
> > asteriskge03*CLI> set verbose 15
> > Verbosity is at least 15
> > asteriskge03*CLI> capi debug
> > CAPI Debugging Enabled
> > asteriskge03*CLI> capi info
> > Contr1: 2 B channels total, 2 B channels free.
> >
> > CONNECT_IND ID=002 #0x2011 LEN=0047
> >   Controller/PLCI/NCCI= 0x101
> >   CIPValue= 0x1
> >   CalledPartyNumber   = 104695467
> >   CallingPartyNumber  = <21 81>108680550
> >   CalledPartySubaddress   = default
> >   CallingPartySubaddress  = default
> >   BC  = <80 90 a3>
> >   LLC = default
> >   HLC = default
> >   AdditionalInfo  = default
> >
> > -- CONNECT_IND
> > (PLCI=0x101,DID=104695467,CID=108680550,CIP=0x1,CONTROLLER=0x1)
> >> BRI1: msn='*' DNID='104695467' MSN
> >   == BRI1: Incoming call '0108680550' -> '104695467'
> > INFO_IND ID=002 #0x2012 LEN=0017
> >   Controller/PLCI/NCCI= 0x101
> >   InfoNumber  = 0x1e
> >   InfoElement = <82 81>
> >
> > INFO_RESP ID=002 #0x2012 LEN=0012
> >   Controller/PLCI/NCCI= 0x101
> >
> > -- BRI1: info

Re: [Asterisk-Users] CHAN_CAPI problem

2006-01-11 Thread asterisk

callerid = 0108680550
--  dialparties.agi: context = macro-dial
--  dialparties.agi: callington = 33
--  dialparties.agi: dnid = 104695467
--  dialparties.agi: request = dialparties.agi
--  dialparties.agi: calleridname = unknown
--  dialparties.agi: extension = s
--  dialparties.agi: language =
--  dialparties.agi: uniqueid = 1136971887.1
--  dialparties.agi: callingpres = 1
--  dialparties.agi: type = CAPI
--  dialparties.agi: rdnis = unknown
--  dialparties.agi: callingtns = 0
--  dialparties.agi: enhanced = 0.0
  dialparties.agi: Caller ID name and number are '0108680550'
  dialparties.agi: Methodology of ring is  'none'
--  dialparties.agi: Added extension 577 to extension map
--  dialparties.agi: Extension 577 cf is disabled
--  dialparties.agi: Extension 577 do not disturb is disabled
   >  dialparties.agi: extnum: 577
   >  dialparties.agi: exthascw: 0
   >  dialparties.agi: exthascfb: 0
   >  dialparties.agi: extcfb:
--  dialparties.agi: Checking CW and CFB status for extension 577
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
--  dialparties.agi: Correct AMPMGRUSER and AMPMGRPASS
  == Manager 'admin' logged off from 127.0.0.1
   >  dialparties.agi: extstate: 0
  dialparties.agi: Extension 577 is available...skipping checks
--  dialparties.agi: DbSet CALLTRACE/577 to 0108680550
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("CAPI/BRI1/104695467-1", "SIP/577|15|tr") in new
stack
-- Called 577
  == BRI1: Requested RINGING-Indication for CAPI/BRI1/104695467-1
ALERT_REQ ID=002 #0x0003 LEN=0018
  Controller/PLCI/NCCI= 0x101
  AdditionalInfo
   BChannelinformation= default
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default
   SendingComplete= default

ALERT_CONF ID=002 #0x0003 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

-- SIP/577-5fc7 is ringing
INFO_IND ID=002 #0x2025 LEN=0017
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8
  InfoElement = <80 90>

INFO_RESP ID=002 #0x2025 LEN=0012
  Controller/PLCI/NCCI= 0x101

-- BRI1: info element CAUSE 80 90
DISCONNECT_IND ID=002 #0x2026 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Reason  = 0x3490

DISCONNECT_RESP ID=002 #0x2026 LEN=0012
  Controller/PLCI/NCCI= 0x101

   > CAPI INFO 0x3490: Normal call clearing
  == Spawn extension (macro-dial, s, 10) exited non-zero on
'CAPI/BRI1/104695467-1' in macro 'dial'
  == Spawn extension (macro-exten-vm, s, 4) exited non-zero on
'CAPI/BRI1/104695467-1' in macro 'exten-vm'
  == Spawn extension (ext-local, 577, 1) exited non-zero on
'CAPI/BRI1/104695467-1'
  == BRI1: CAPI Hangingup
  == BRI1: Interface cleanup PLCI=0x101
   > CAPI devicestate requested for BRI1/104695467
   > CAPI devicestate requested for BRI1/104695467
***

What is the meaning of immediate=yes ? I read the comment , but I was not
able to understand it;
In other words: in a ;'MSN' (point-to-multipoint)  (not DID) configuration,
it is always required "immediate=yes" or is it due to a particular
configuration of my Telecom ISDN BRI ? If it is always required, I could
suggest you to put a comment about this on the capi.conf..

Thank you very much again,

Andrea






   
 Armin Schindler       
 <[EMAIL PROTECTED] 
 >  To 
 Sent by:  Asterisk Users Mailing List -   
 asterisk-users-bo Non-Commercial Discussion   
 [EMAIL PROTECTED]
 m.com  cc 
   
   Subject 
 11/01/2006 10.24  Re: [Asterisk-Users] CHAN_CAPI  
   problem 
   
 Please respond to 
  Asterisk Users   
  Mailing List -

Re: [Asterisk-Users] CHAN_CAPI problem

2006-01-11 Thread Armin Schindler

There is no 'sending-complete'/'setup' info-element, please use
immediate=yes in capi.conf

Armin


On Wed, 11 Jan 2006 [EMAIL PROTECTED] wrote:
> Thank you very much for your attention;
> Here is what you asked for:
> ***
> asteriskge03*CLI> set verbose 15
> Verbosity is at least 15
> asteriskge03*CLI> capi debug
> CAPI Debugging Enabled
> asteriskge03*CLI> capi info
> Contr1: 2 B channels total, 2 B channels free.
> 
> CONNECT_IND ID=002 #0x2011 LEN=0047
>   Controller/PLCI/NCCI= 0x101
>   CIPValue= 0x1
>   CalledPartyNumber   = 104695467
>   CallingPartyNumber  = <21 81>108680550
>   CalledPartySubaddress   = default
>   CallingPartySubaddress  = default
>   BC  = <80 90 a3>
>   LLC = default
>   HLC = default
>   AdditionalInfo  = default
> 
> -- CONNECT_IND
> (PLCI=0x101,DID=104695467,CID=108680550,CIP=0x1,CONTROLLER=0x1)
>> BRI1: msn='*' DNID='104695467' MSN
>   == BRI1: Incoming call '0108680550' -> '104695467'
> INFO_IND ID=002 #0x2012 LEN=0017
>   Controller/PLCI/NCCI= 0x101
>   InfoNumber  = 0x1e
>   InfoElement = <82 81>
> 
> INFO_RESP ID=002 #0x2012 LEN=0012
>   Controller/PLCI/NCCI= 0x101
> 
> -- BRI1: info element PI 82 81
>> BRI1: Not end-to-end ISDN
> INFO_IND ID=002 #0x2013 LEN=0025
>   Controller/PLCI/NCCI= 0x101
>   InfoNumber  = 0x70
>   InfoElement = 104695467
> 
> INFO_RESP ID=002 #0x2013 LEN=0012
>   Controller/PLCI/NCCI= 0x101
> 
> -- BRI1: info element CALLED PARTY NUMBER
>> BRI1: INFO_IND DID digits not used in this state.
> INFO_IND ID=002 #0x2014 LEN=0016
>   Controller/PLCI/NCCI= 0x101
>   InfoNumber  = 0x18
>   InfoElement = <89>
> 
> INFO_RESP ID=002 #0x2014 LEN=0012
>   Controller/PLCI/NCCI= 0x101
> 
> -- BRI1: info element CHANNEL IDENTIFICATION 89
> DISCONNECT_IND ID=002 #0x2017 LEN=0014
>   Controller/PLCI/NCCI= 0x101
>   Reason  = 0x0
> 
> DISCONNECT_RESP ID=002 #0x2017 LEN=0012
>   Controller/PLCI/NCCI= 0x101
> 
> -- BRI1: DISCONNECT_IND on incoming without pbx, doing hangup.
>   == BRI1: CAPI Hangingup
>   == BRI1: Interface cleanup PLCI=0x101
>> CAPI devicestate requested for BRI1/104695467
> 
> ***
> 
> The lines:
> 
> BRI1: DISCONNECT_IND on incoming without pbx, doing hangup.
>   == BRI1: CAPI Hangingup
>   == BRI1: Interface cleanup PLCI=0x101
>> CAPI devicestate requested for BRI1/104695467
> 
> appeared on the console WHILE I was still earing the ring tone on the
> calling phone. When I , at last, after other 4 rings, hangup the calling
> phone,
> nothing changed on the console
> 
> 
> Andrea
> 
> 
> 
>
>  Armin Schindler   
>  <[EMAIL PROTECTED] 
>  >  To 
>  Sent by:  Asterisk Users Mailing List -   
>  asterisk-users-bo     Non-Commercial Discussion   
>  [EMAIL PROTECTED]
>  m.com  cc 
>
>Subject 
>  10/01/2006 19.16  Re: [Asterisk-Users] CHAN_CAPI  
>problem 
>
>  Please respond to 
>   Asterisk Users   
>   Mailing List -   
>   Non-Commercial   
> Discussion 
>  <[EMAIL PROTECTED]

Re: [Asterisk-Users] CHAN_CAPI problem

2006-01-11 Thread asterisk

Thank you very much for your attention;
Here is what you asked for:
***
asteriskge03*CLI> set verbose 15
Verbosity is at least 15
asteriskge03*CLI> capi debug
CAPI Debugging Enabled
asteriskge03*CLI> capi info
Contr1: 2 B channels total, 2 B channels free.

CONNECT_IND ID=002 #0x2011 LEN=0047
  Controller/PLCI/NCCI= 0x101
  CIPValue= 0x1
  CalledPartyNumber   = 104695467
  CallingPartyNumber  = <21 81>108680550
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BC  = <80 90 a3>
  LLC = default
  HLC = default
  AdditionalInfo  = default

-- CONNECT_IND
(PLCI=0x101,DID=104695467,CID=108680550,CIP=0x1,CONTROLLER=0x1)
   > BRI1: msn='*' DNID='104695467' MSN
  == BRI1: Incoming call '0108680550' -> '104695467'
INFO_IND ID=002 #0x2012 LEN=0017
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x1e
  InfoElement = <82 81>

INFO_RESP ID=002 #0x2012 LEN=0012
  Controller/PLCI/NCCI= 0x101

-- BRI1: info element PI 82 81
   > BRI1: Not end-to-end ISDN
INFO_IND ID=002 #0x2013 LEN=0025
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x70
  InfoElement = 104695467

INFO_RESP ID=002 #0x2013 LEN=0012
  Controller/PLCI/NCCI= 0x101

-- BRI1: info element CALLED PARTY NUMBER
   > BRI1: INFO_IND DID digits not used in this state.
INFO_IND ID=002 #0x2014 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = <89>

INFO_RESP ID=002 #0x2014 LEN=0012
  Controller/PLCI/NCCI= 0x101

-- BRI1: info element CHANNEL IDENTIFICATION 89
DISCONNECT_IND ID=002 #0x2017 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Reason  = 0x0

DISCONNECT_RESP ID=002 #0x2017 LEN=0012
  Controller/PLCI/NCCI= 0x101

-- BRI1: DISCONNECT_IND on incoming without pbx, doing hangup.
  == BRI1: CAPI Hangingup
  == BRI1: Interface cleanup PLCI=0x101
   > CAPI devicestate requested for BRI1/104695467

***

The lines:

BRI1: DISCONNECT_IND on incoming without pbx, doing hangup.
  == BRI1: CAPI Hangingup
  == BRI1: Interface cleanup PLCI=0x101
   > CAPI devicestate requested for BRI1/104695467

appeared on the console WHILE I was still earing the ring tone on the
calling phone. When I , at last, after other 4 rings, hangup the calling
phone,
nothing changed on the console


Andrea



   
 Armin Schindler   
 <[EMAIL PROTECTED] 
 >  To 
 Sent by:  Asterisk Users Mailing List -   
 asterisk-users-bo Non-Commercial Discussion   
 [EMAIL PROTECTED]
 m.com  cc 
   
           Subject 
 10/01/2006 19.16  Re: [Asterisk-Users] CHAN_CAPI  
   problem 
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 <[EMAIL PROTECTED] 
 ists.digium.com>  
   
   




On Tue, 10 Jan 2006 [EMAIL PROTECTED] wrote:
> Thank you.
> I already downloaded and installed it (they are dated 07-01-2006, version
> 0.6.3, correct ?)

Yes.

> I maked clean, make and make install.
>
> Nothing changed, dial out perfect, dial in: (capi debug on)
>
> asteriskge03*CLI> capi info
> Contr1: 2 B channels total, 2 B channels free.
> asteriskge03*CLI>
> asteriskge03*CLI>
> -

Re: [Asterisk-Users] CHAN_CAPI problem

2006-01-10 Thread Armin Schindler

On Tue, 10 Jan 2006 [EMAIL PROTECTED] wrote:
> Thank you.
> I already downloaded and installed it (they are dated 07-01-2006, version
> 0.6.3, correct ?)

Yes.

> I maked clean, make and make install.
> 
> Nothing changed, dial out perfect, dial in: (capi debug on)
> 
> asteriskge03*CLI> capi info
> Contr1: 2 B channels total, 2 B channels free.
> asteriskge03*CLI>
> asteriskge03*CLI>
> -- CONNECT_IND
> (PLCI=0x101,DID=104695467,CID=108680550,CIP=0x1,CONTROLLER=0x1)
>   == BRI1: Interface cleanup PLCI=0x101
> 
> BRI1 is the name of my interface
> 
> could it be a kernel issue ?? I am using SUSE Linux 10;

I don't think so. Please increase the the verbose level to 5
(set verbose 5) in addition to 'capi debug'.

Armin
 
> kernel :  2.6.13-15.7-smp
> 
> Andrea
> 
> 
> 
>
>  Armin Schindler   
>  <[EMAIL PROTECTED] 
>  >  To 
>  Sent by:  Asterisk Users Mailing List -   
>  asterisk-users-bo Non-Commercial Discussion   
>  [EMAIL PROTECTED]
>  m.com  cc 
>
>    Subject 
>  10/01/2006 18.29  Re: [Asterisk-Users] CHAN_CAPI  
>problem 
>
>  Please respond to 
>   Asterisk Users   
>   Mailing List -   
>   Non-Commercial   
> Discussion 
>  <[EMAIL PROTECTED] 
>  ists.digium.com>  
>
>
> 
> 
> 
> 
> I suggest you use the newer chan_capi-cm (loadable from sourceforge.net).
> 
> Armin
> 
> On Tue, 10 Jan 2006 [EMAIL PROTECTED] wrote:
> > Hi all,
> > I installed asterisk stable cvs 1.2 and chan_capi 0.4.0 PRE1, with one
> AVM
> > Fritz Card ISDN connected to a Telecom NT1 Plus
> >
> > I configured asterisk via AMP.
> > No problem in making calls.
> > If I try to ring the ISDN Phone Number, I don't see anything on the
> > asterisk Console,
> > I I activate the capi debug , I see the ring on the capi channel.
> > If the context were wrong , I anyway should  see some line about this
> >
> > Why I cannot see anything on asterisk ,nor in the /var/log/asterisk/full
> ?
> >
> > here is my /etc/asterisk/capi.conf
> >
> > asteriskge03:/etc/asterisk # cat capi.conf
> > ;
> > ; CAPI config
> > ;
> > ;
> >
> > ; general section
> >
> > [general]
> > nationalprefix=0
> > internationalprefix=00
> > rxgain=0.8
> > txgain=0.8
> > ;ulaw=yes;set this, if you live in u-law world instead of a-law
> >
> > ; interface sections ...
> >
> > [BRI1]  ;this example interface gets name 'ISDN1' and may be any
> >  ;name not starting with 'g' or 'contr'.
> > ;ntmode=yes  ;if isdn card operates in nt mode, set this to yes
> > isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward
> dial)
> >  ;when using NT-mode, 'DID' should be set in any case
> > incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any
> > ;defaultcid=123  ;set a default caller id to that interface for dial-out,
> >  ;this caller id will be used when dial option 'd' is
> set.
> > ;controller=0;ISDN4BSD default
> > ;controller=7;ISDN4BSD USB default
> > controller=1 ;capi controller number to use
> > group=1  ;dialout group
> > ;prefix=0;set a prefix to calling number on incoming calls
> > softdtmf=on  ;enable/disable software dtmf detection, rec

Re: [Asterisk-Users] CHAN_CAPI problem

2006-01-10 Thread asterisk

Thank you.
I already downloaded and installed it (they are dated 07-01-2006, version
0.6.3, correct ?)
I maked clean, make and make install.

Nothing changed, dial out perfect, dial in: (capi debug on)

asteriskge03*CLI> capi info
Contr1: 2 B channels total, 2 B channels free.
asteriskge03*CLI>
asteriskge03*CLI>
-- CONNECT_IND
(PLCI=0x101,DID=104695467,CID=108680550,CIP=0x1,CONTROLLER=0x1)
  == BRI1: Interface cleanup PLCI=0x101

BRI1 is the name of my interface

could it be a kernel issue ?? I am using SUSE Linux 10;

kernel :  2.6.13-15.7-smp

Andrea



   
 Armin Schindler   
 <[EMAIL PROTECTED] 
 >  To 
 Sent by:  Asterisk Users Mailing List -   
 asterisk-users-bo Non-Commercial Discussion   
 [EMAIL PROTECTED]
 m.com  cc 
   
   Subject 
 10/01/2006 18.29  Re: [Asterisk-Users] CHAN_CAPI  
   problem 
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 <[EMAIL PROTECTED] 
 ists.digium.com>  
   
   




I suggest you use the newer chan_capi-cm (loadable from sourceforge.net).

Armin

On Tue, 10 Jan 2006 [EMAIL PROTECTED] wrote:
> Hi all,
> I installed asterisk stable cvs 1.2 and chan_capi 0.4.0 PRE1, with one
AVM
> Fritz Card ISDN connected to a Telecom NT1 Plus
>
> I configured asterisk via AMP.
> No problem in making calls.
> If I try to ring the ISDN Phone Number, I don't see anything on the
> asterisk Console,
> I I activate the capi debug , I see the ring on the capi channel.
> If the context were wrong , I anyway should  see some line about this
>
> Why I cannot see anything on asterisk ,nor in the /var/log/asterisk/full
?
>
> here is my /etc/asterisk/capi.conf
>
> asteriskge03:/etc/asterisk # cat capi.conf
> ;
> ; CAPI config
> ;
> ;
>
> ; general section
>
> [general]
> nationalprefix=0
> internationalprefix=00
> rxgain=0.8
> txgain=0.8
> ;ulaw=yes;set this, if you live in u-law world instead of a-law
>
> ; interface sections ...
>
> [BRI1]  ;this example interface gets name 'ISDN1' and may be any
>  ;name not starting with 'g' or 'contr'.
> ;ntmode=yes  ;if isdn card operates in nt mode, set this to yes
> isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward
dial)
>  ;when using NT-mode, 'DID' should be set in any case
> incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any
> ;defaultcid=123  ;set a default caller id to that interface for dial-out,
>  ;this caller id will be used when dial option 'd' is
set.
> ;controller=0;ISDN4BSD default
> ;controller=7;ISDN4BSD USB default
> controller=1 ;capi controller number to use
> group=1  ;dialout group
> ;prefix=0;set a prefix to calling number on incoming calls
> softdtmf=on  ;enable/disable software dtmf detection, recommended for
> AVM cards
> relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
> detection
> accountcode= ;Asterisk accountcode to use in CDRs
> context=from-pstn  ;context for incoming calls
> holdtype=hold;when Asterisk puts the call on hold, ISDN HOLD will be
> used. If
>  ;set to 'local' (default value), no hold is done and
> Asterisk may
>  ;play MOH.
> ;immediate=yes   ;DID: immediate start of pbx with extension 's' if no
> digits were
>  ; received on incoming call (no destination number
> yet)
>  ;MSN: start pbx on CONNECT_IND and don

Re: [Asterisk-Users] CHAN_CAPI problem

2006-01-10 Thread Armin Schindler

I suggest you use the newer chan_capi-cm (loadable from sourceforge.net).

Armin

On Tue, 10 Jan 2006 [EMAIL PROTECTED] wrote:
> Hi all,
> I installed asterisk stable cvs 1.2 and chan_capi 0.4.0 PRE1, with one AVM
> Fritz Card ISDN connected to a Telecom NT1 Plus
> 
> I configured asterisk via AMP.
> No problem in making calls.
> If I try to ring the ISDN Phone Number, I don't see anything on the
> asterisk Console,
> I I activate the capi debug , I see the ring on the capi channel.
> If the context were wrong , I anyway should  see some line about this
> 
> Why I cannot see anything on asterisk ,nor in the /var/log/asterisk/full  ?
> 
> here is my /etc/asterisk/capi.conf
> 
> asteriskge03:/etc/asterisk # cat capi.conf
> ;
> ; CAPI config
> ;
> ;
> 
> ; general section
> 
> [general]
> nationalprefix=0
> internationalprefix=00
> rxgain=0.8
> txgain=0.8
> ;ulaw=yes;set this, if you live in u-law world instead of a-law
> 
> ; interface sections ...
> 
> [BRI1]  ;this example interface gets name 'ISDN1' and may be any
>  ;name not starting with 'g' or 'contr'.
> ;ntmode=yes  ;if isdn card operates in nt mode, set this to yes
> isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
>  ;when using NT-mode, 'DID' should be set in any case
> incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any
> ;defaultcid=123  ;set a default caller id to that interface for dial-out,
>  ;this caller id will be used when dial option 'd' is set.
> ;controller=0;ISDN4BSD default
> ;controller=7;ISDN4BSD USB default
> controller=1 ;capi controller number to use
> group=1  ;dialout group
> ;prefix=0;set a prefix to calling number on incoming calls
> softdtmf=on  ;enable/disable software dtmf detection, recommended for
> AVM cards
> relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
> detection
> accountcode= ;Asterisk accountcode to use in CDRs
> context=from-pstn  ;context for incoming calls
> holdtype=hold;when Asterisk puts the call on hold, ISDN HOLD will be
> used. If
>  ;set to 'local' (default value), no hold is done and
> Asterisk may
>  ;play MOH.
> ;immediate=yes   ;DID: immediate start of pbx with extension 's' if no
> digits were
>  ; received on incoming call (no destination number
> yet)
>  ;MSN: start pbx on CONNECT_IND and don't wait for
> SETUP/SENDING-COMPLETE.
>  ; info like REDIRECTINGNUMBER may be lost, but this is
> necessary for
>  ; drivers/pbx/telco which does not send SETUP or
> SENDING-COMPLETE.
> ;echosquelch=1   ;_VERY_PRIMITIVE_ echo suppression
> ;echocancel=yes  ;EICON DIVA SERVER (CAPI) echo cancelation
>  ;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
> echocancelold=yes;use facility selector 6 instead of correct 8 (necessary
> for older eicon drivers)
> ;echotail=64 ;echo cancel tail setting
> ;bridge=yes  ;native bridging (CAPI line interconnect) if available
> ;callgroup=1 ;Asterisk call group
> devices=2;number of concurrent calls on this controller
>  ;(2 makes sense for single BRI, 30 for PRI)
> 
> thanks in advance,
> Andrea
> 
> Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.
> 
> Visitate il sito http://www.frameweb.it
> 
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[Asterisk-Users] CHAN_CAPI problem

2006-01-10 Thread asterisk

Hi all,
I installed asterisk stable cvs 1.2 and chan_capi 0.4.0 PRE1, with one AVM
Fritz Card ISDN connected to a Telecom NT1 Plus

I configured asterisk via AMP.
No problem in making calls.
If I try to ring the ISDN Phone Number, I don't see anything on the
asterisk Console,
I I activate the capi debug , I see the ring on the capi channel.
If the context were wrong , I anyway should  see some line about this

Why I cannot see anything on asterisk ,nor in the /var/log/asterisk/full  ?

here is my /etc/asterisk/capi.conf

asteriskge03:/etc/asterisk # cat capi.conf
;
; CAPI config
;
;

; general section

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
;ulaw=yes;set this, if you live in u-law world instead of a-law

; interface sections ...

[BRI1]  ;this example interface gets name 'ISDN1' and may be any
 ;name not starting with 'g' or 'contr'.
;ntmode=yes  ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
 ;when using NT-mode, 'DID' should be set in any case
incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any
;defaultcid=123  ;set a default caller id to that interface for dial-out,
 ;this caller id will be used when dial option 'd' is set.
;controller=0;ISDN4BSD default
;controller=7;ISDN4BSD USB default
controller=1 ;capi controller number to use
group=1  ;dialout group
;prefix=0;set a prefix to calling number on incoming calls
softdtmf=on  ;enable/disable software dtmf detection, recommended for
AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
detection
accountcode= ;Asterisk accountcode to use in CDRs
context=from-pstn  ;context for incoming calls
holdtype=hold;when Asterisk puts the call on hold, ISDN HOLD will be
used. If
 ;set to 'local' (default value), no hold is done and
Asterisk may
 ;play MOH.
;immediate=yes   ;DID: immediate start of pbx with extension 's' if no
digits were
 ; received on incoming call (no destination number
yet)
 ;MSN: start pbx on CONNECT_IND and don't wait for
SETUP/SENDING-COMPLETE.
 ; info like REDIRECTINGNUMBER may be lost, but this is
necessary for
 ; drivers/pbx/telco which does not send SETUP or
SENDING-COMPLETE.
;echosquelch=1   ;_VERY_PRIMITIVE_ echo suppression
;echocancel=yes  ;EICON DIVA SERVER (CAPI) echo cancelation
 ;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
echocancelold=yes;use facility selector 6 instead of correct 8 (necessary
for older eicon drivers)
;echotail=64 ;echo cancel tail setting
;bridge=yes  ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;Asterisk call group
devices=2;number of concurrent calls on this controller
 ;(2 makes sense for single BRI, 30 for PRI)

thanks in advance,
Andrea

Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

Visitate il sito http://www.frameweb.it

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Re: [Asterisk-Users] chan_capi-cm 0.6.1 won't load

2005-12-22 Thread Johan Helsingius

> Please use latest sources from CVS on sourceforge for chan_capi-cm

Seems to have helped! Thanks!

Julf
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Re: [Asterisk-Users] chan_capi-cm 0.6.1 won't load

2005-12-22 Thread Armin Schindler

On Wed, 21 Dec 2005, Johan Helsingius wrote:
> Asterisk 1.2.1 on gentoo. Trying to use chan_capi-cm 0.6.1
> results in:
> 
> WARNING[11724] loader.c: /usr/lib/asterisk/modules/chan_capi.so:
> undefined symbol: use_ast_mutex_init_instead_of_pthread_mutex_init
> WARNING[11724] loader.c: Loading module chan_capi.so failed!
> 
> Sounds like some version incompatibility - any ideas?

Please use latest sources from CVS on sourceforge for chan_capi-cm

Armin

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[Asterisk-Users] chan_capi-cm 0.6.1 won't load

2005-12-21 Thread Johan Helsingius

Asterisk 1.2.1 on gentoo. Trying to use chan_capi-cm 0.6.1
results in:

WARNING[11724] loader.c: /usr/lib/asterisk/modules/chan_capi.so:
undefined symbol: use_ast_mutex_init_instead_of_pthread_mutex_init
WARNING[11724] loader.c: Loading module chan_capi.so failed!

Sounds like some version incompatibility - any ideas?

Julf
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Re: [Asterisk-Users] chan_capi AVM C2

2005-12-14 Thread Armin Schindler

On Wed, 14 Dec 2005, stéphane plichon wrote:
> Armin Schindler wrote:
> > On Wed, 14 Dec 2005, stéphane plichon wrote:
> > 
> context=capi-in
> devices=2
> >>>
> >>>
> >>>This is just one section which two sets of options. You need to define two 
> >>>sections with [...]. See README.
> >>>
> >>>Armin
> >>>
> >>>
> >>
> >>no in or out call if i do that (with or without [interfaces]):
> >>
> >>[general]
> >>nationalprefix=0
> >>internationalprefix=00
> >>rxgain=0.8
> >>txgain=0.8
> >>
> >>;[interfaces]
> >>
> >>[contr1]
> >>blah...
> >>
> >>[contr2]
> >>blah
> >>
> >>in readme i don't read section [] for each controller
> >>can you post your capi.conf plz ?
> > 
> > 
> > Send me a verbose log level 5 with 'capi debug', if the following
> > does not work.
> > 
> > Armin
> > 
> nothing, tut-tut-tut signal nothing in the log

If you don't get anything from capi log with 'capi debug' and 'set verbose 5'
then maybe your extensions.conf is wrong.

Armin
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Re: [Asterisk-Users] chan_capi AVM C2

2005-12-14 Thread stéphane plichon

Armin Schindler wrote:
> On Wed, 14 Dec 2005, stéphane plichon wrote:
> 
context=capi-in
devices=2
>>>
>>>
>>>This is just one section which two sets of options. You need to define two 
>>>sections with [...]. See README.
>>>
>>>Armin
>>>
>>>
>>
>>no in or out call if i do that (with or without [interfaces]):
>>
>>[general]
>>nationalprefix=0
>>internationalprefix=00
>>rxgain=0.8
>>txgain=0.8
>>
>>;[interfaces]
>>
>>[contr1]
>>blah...
>>
>>[contr2]
>>blah
>>
>>in readme i don't read section [] for each controller
>>can you post your capi.conf plz ?
> 
> 
> Send me a verbose log level 5 with 'capi debug', if the following
> does not work.
> 
> Armin
> 
nothing, tut-tut-tut signal nothing in the log


-- 
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Re: [Asterisk-Users] chan_capi AVM C2

2005-12-14 Thread Armin Schindler

On Wed, 14 Dec 2005, stéphane plichon wrote:
> >>context=capi-in
> >>devices=2
> > 
> > 
> > This is just one section which two sets of options. You need to define two 
> > sections with [...]. See README.
> > 
> > Armin
> > 
> > 
> 
> no in or out call if i do that (with or without [interfaces]):
> 
> [general]
> nationalprefix=0
> internationalprefix=00
> rxgain=0.8
> txgain=0.8
> 
> ;[interfaces]
> 
> [contr1]
> blah...
> 
> [contr2]
> blah
> 
> in readme i don't read section [] for each controller
> can you post your capi.conf plz ?

Send me a verbose log level 5 with 'capi debug', if the following
does not work.

Armin

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
 
[AVM1]
isdnmode=msn
incomingmsn=*
controller=1
context=capi-in
devices=2

[AVM2]
isdnmode=msn
incomingmsn=*
controller=2
context=capi-in
devices=2

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Re: [Asterisk-Users] chan_capi AVM C2

2005-12-14 Thread stéphane plichon


>>context=capi-in
>>devices=2
> 
> 
> This is just one section which two sets of options. You need to define two 
> sections with [...]. See README.
> 
> Armin
> 
> 

no in or out call if i do that (with or without [interfaces]):

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

;[interfaces]

[contr1]
blah...

[contr2]
blah

in readme i don't read section [] for each controller
can you post your capi.conf plz ?


-- 
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jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] chan_capi AVM C2

2005-12-13 Thread Armin Schindler

On Wed, 14 Dec 2005, stéphane plichon wrote:
> stéphane plichon wrote:
> > stéphane plichon wrote:
> > 
> >>Hi all,
> >>
> >>currently i running * 1.0.9 with chan_capi 0.3.5
> >>
> >>my first problem is:
> >>
> >>in incoming call, when BCHAN is full in contr1 incoming call on contr2
> >>are not answered with error :
> >>
> >>chan_capi.c:1953 in capi_handle_msg: received a call waiting CONNECT_IN
> >>
> >>but if use different msn in capi.conf incoming call works on both controler
> >>
> >>
> >>
> > 
> > ok, now working, but i get only ring on third call
> > 
> 
> always 2 sections, i'm going to upgrade
> 
> ;
> ; CAPI config
> ;
> ;
> [general]
> nationalprefix=0
> internationalprefix=00
> rxgain=0.8
> txgain=0.8
> 
> [interfaces]
> 
> isdnmode=ptp
> ;msn=9881
> incomingmsn=* ;9881,2942,2943,2944,2945,2946,2947,2948,2949
> controller=1
> group=1
> deflect=0472529256
> softdtmf=1
> accountcode=
> context=capi-in
> devices=2
> 
> isdnmode=ptp
> ;msn=9881
> incomingmsn=* ;9881,2942,2943,2944,2945,2946,2947,2948,2949
> controller=2
> deflect=0472529256
> group=1
> softdtmf=1
> accountcode=
> context=capi-in
> devices=2

This is just one section which two sets of options. You need to define two 
sections with [...]. See README.

Armin
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Re: [Asterisk-Users] chan_capi AVM C2

2005-12-13 Thread stéphane plichon

stéphane plichon wrote:
> stéphane plichon wrote:
> 
>>Hi all,
>>
>>currently i running * 1.0.9 with chan_capi 0.3.5
>>
>>my first problem is:
>>
>>in incoming call, when BCHAN is full in contr1 incoming call on contr2
>>are not answered with error :
>>
>>chan_capi.c:1953 in capi_handle_msg: received a call waiting CONNECT_IN
>>
>>but if use different msn in capi.conf incoming call works on both controler
>>
>>
>>
> 
> ok, now working, but i get only ring on third call
> 

always 2 sections, i'm going to upgrade

;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]

isdnmode=ptp
;msn=9881
incomingmsn=* ;9881,2942,2943,2944,2945,2946,2947,2948,2949
controller=1
group=1
deflect=0472529256
softdtmf=1
accountcode=
context=capi-in
devices=2

isdnmode=ptp
;msn=9881
incomingmsn=* ;9881,2942,2943,2944,2945,2946,2947,2948,2949
controller=2
deflect=0472529256
group=1
softdtmf=1
accountcode=
context=capi-in
devices=2


-- 
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tel: +33 (0)472529881
fax: +33 (0)472177764
web: http://www.hasgard.net
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Re: [Asterisk-Users] chan_capi AVM C2

2005-12-13 Thread Michael J. Tubby G8TIC


Recommend you upgrade to Asterisk 1.2.1 and chan_capi-cm-0.6.1

Have no problems here with a AVM C4 and 2 lines (4 channels in P2P mode) 
plus a line in P2MP (MSN) mode.


Mike

- Original Message - 
From: "stéphane plichon" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Tuesday, December 13, 2005 5:09 PM
Subject: Re: [Asterisk-Users] chan_capi AVM C2


stéphane plichon wrote:

Hi all,

currently i running * 1.0.9 with chan_capi 0.3.5

my first problem is:

in incoming call, when BCHAN is full in contr1 incoming call on contr2
are not answered with error :

chan_capi.c:1953 in capi_handle_msg: received a call waiting CONNECT_IN

but if use different msn in capi.conf incoming call works on both 
controler





ok, now working, but i get only ring on third call

--
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Re: [Asterisk-Users] chan_capi AVM C2

2005-12-13 Thread Armin Schindler

On Tue, 13 Dec 2005, stéphane plichon wrote:
> stéphane plichon wrote:
> > Hi all,
> > 
> > currently i running * 1.0.9 with chan_capi 0.3.5
> > 
> > my first problem is:
> > 
> > in incoming call, when BCHAN is full in contr1 incoming call on contr2
> > are not answered with error :
> > 
> > chan_capi.c:1953 in capi_handle_msg: received a call waiting CONNECT_IN
> > 
> > but if use different msn in capi.conf incoming call works on both controler
> > 
> > 
> > 
> ok, now working, but i get only ring on third call

Make sure you have 2 separate sections in your capi.conf, one for each 
controller.

Armin
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Re: [Asterisk-Users] chan_capi AVM C2

2005-12-13 Thread stéphane plichon

stéphane plichon wrote:
> Hi all,
> 
> currently i running * 1.0.9 with chan_capi 0.3.5
> 
> my first problem is:
> 
> in incoming call, when BCHAN is full in contr1 incoming call on contr2
> are not answered with error :
> 
> chan_capi.c:1953 in capi_handle_msg: received a call waiting CONNECT_IN
> 
> but if use different msn in capi.conf incoming call works on both controler
> 
> 
> 
ok, now working, but i get only ring on third call

-- 
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Re: [Asterisk-Users] chan_capi AVM C2

2005-12-13 Thread stéphane plichon

Philipp von Klitzing wrote:
> Hi!
> 
> 
>>currently i running * 1.0.9 with chan_capi 0.3.5
> 
> 
> Try chan_capi-cm instead and see if it helps.
> 
> Cheers, Philipp
> 
> 
> 
compiling 0.5.4 when there was more than 2 call i got :
 ERROR[6060]: chan_capi.c:2324 capi_handle_connect_indication: received
a call waiting CONNECT_IND

:-(


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Re: [Asterisk-Users] chan_capi AVM C2

2005-12-13 Thread Philipp von Klitzing

Hi!

> currently i running * 1.0.9 with chan_capi 0.3.5

Try chan_capi-cm instead and see if it helps.

Cheers, Philipp



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[Asterisk-Users] chan_capi AVM C2

2005-12-13 Thread stéphane plichon

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi all,

currently i running * 1.0.9 with chan_capi 0.3.5

my first problem is:

in incoming call, when BCHAN is full in contr1 incoming call on contr2
are not answered with error :

chan_capi.c:1953 in capi_handle_msg: received a call waiting CONNECT_IN

but if use different msn in capi.conf incoming call works on both controler



- --
Stephane Plichon | HASGARD
jabber: [EMAIL PROTECTED]
~
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)

iD8DBQFDnpAYMI/jEEfAy/4RAgqhAJ9w7x+org8dQtiK2Ke5E3NPBg2AeQCfVAos
2uO9vsdVaZDvt9zK4H2X9uU=
=ptnp
-END PGP SIGNATURE-
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Re: [Asterisk-Users] chan_capi fails when Asterisk doesn't start under root user

2005-11-17 Thread Armin Schindler

On Thu, 17 Nov 2005, Amaury BOSSE wrote:
> I have a problem with chan_capi-cm-0.6.1.
> 
> It works when I start * directly from the command line with "asterisk
> -vvvgc", but not when I use "/etc/init.d/asterisk" script.
> 
>  
> 
> I have the log bellow in /var/log/asterisk/messages :
> 
>  
> 
> Nov 17 11:26:43 WARNING[5337]: CAPI not installed, CAPI disabled!
> 
> Nov 17 11:26:43 WARNING[5337]: chan_capi.so: load_module failed, returning
> -1
> 
> Nov 17 11:26:43 WARNING[5337]: Loading module chan_capi.so failed!
> 
>  
> 
> I have changed parameters in the starting script in order to start * as root
> instead of as asterisk and It seems to works.

Maybe your permissions on /dev/capi20 are wrong?

Armin

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[Asterisk-Users] chan_capi fails when Asterisk doesn't start under root user

2005-11-17 Thread Amaury BOSSE









I have a problem with chan_capi-cm-0.6.1.

It works when I start * directly from the command
line with “asterisk –vvvgc”, but not when I use
“/etc/init.d/asterisk” script.

 

I have the log bellow in /var/log/asterisk/messages :

 

Nov 17 11:26:43 WARNING[5337]: CAPI not installed,
CAPI disabled!

Nov 17 11:26:43 WARNING[5337]: chan_capi.so:
load_module failed, returning -1

Nov 17 11:26:43 WARNING[5337]: Loading module
chan_capi.so failed!

 

I have changed parameters in the starting script in
order to start * as root instead of as asterisk and It seems to works.

 

Is someone already had this problem or knows where
does it come from.

I have search the web to find answers but without
results.

 

Thanks






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Re: AW: [Asterisk-Users] chan_capi-cm-0.6 can't be loaded with latest asterisk version from cvs

2005-11-12 Thread Armin Schindler

On Sat, 12 Nov 2005, Faris Raouf wrote:
> [EMAIL PROTECTED] wrote:
> > Thanks Armin, this version is working, but I still have an undefined
> > symbol
> > in another module:
> > 
> > 
> > [pbx_wilcalu.so]Nov  5 18:51:12 WARNING[11348]: loader.c:325
> > __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined
> > symbol:
> > ast_pthread_create
> > Nov  5 18:51:12 WARNING[11348]: loader.c:554 load_modules: Loading module
> > pbx_wilcalu.so failed!
> > 
> > Can you also help me on that issue?
> > 
> > Thanks and Regards
> > 
> > Markus
> > 
> 
> To my knowledge, that module has nothing to do with CAPI. I don't honestly
> know what it does. ("will call you")
> 
> What I can say is that with 1.2 RC2 (latest from CVS) and chan_capi-cd 0.6
> (latest from sourceforge cvs) as of 20:00 12/11/05 GMT on RedHat 9, I get
> exactly the same error when loading on a freshly sanitised system with all
> traces of previous asterisk installations removed.
> 
> HOWEVER, if you add a noload => pbx_wilcalu.so in modules.conf you can make
> the error go away. (but this is probably a bad thing since I don't know what
> that module does!)
> 
> But unfortunately, for me at least, I then end up with errors about:
> 
> app_capiCD.so
> app_capiHOLD.so
> app_capiRETRIEVE.so
> app_capiECT.so
> and
> app_capiMCID.so
> 
> 
> For example:
> 
> [app_capiCD.so]Nov 12 20:24:55 WARNING[19197]: loader.c:325 __load_resource:
> /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol:
> ast_capi_MessageNumber
> Nov 12 20:24:55 WARNING[19197]: loader.c:554 load_modules: Loading module
> app_capiCD.so failed!
> 
> # Ouch ... error while writing audio data: : Broken pipe
> 
> No matter which of the modules you comment out above, the same thing happens
> -- the error is always about app_capi_MessageNumber
> 
> Armin (or anybody) -- have I missed something out/done something wrong, or is
> it a compatibility issue between chan_capi-cm 0.6 and Asterisk 1.2 RC2?

I cannot tell abything about the pbx_wilcalu.so issue, but with current
chan_capi-cm all app_capi* modules are obsolete and may not be used any
more. Just have a look at the chan_capi-cm package/cvs-contents, them
modules are removed, so why are you trying to load these?
As stated in README, the functionality is not part of chan_capi.so itself.

Armin

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Re: AW: [Asterisk-Users] chan_capi-cm-0.6 can't be loaded with latest asterisk version from cvs

2005-11-12 Thread Faris Raouf


[EMAIL PROTECTED] wrote:

Thanks Armin, this version is working, but I still have an undefined symbol
in another module:


[pbx_wilcalu.so]Nov  5 18:51:12 WARNING[11348]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol:
ast_pthread_create
Nov  5 18:51:12 WARNING[11348]: loader.c:554 load_modules: Loading module
pbx_wilcalu.so failed!

Can you also help me on that issue?

Thanks and Regards

Markus



To my knowledge, that module has nothing to do with CAPI. I don't 
honestly know what it does. ("will call you")


What I can say is that with 1.2 RC2 (latest from CVS) and chan_capi-cd 
0.6 (latest from sourceforge cvs) as of 20:00 12/11/05 GMT on RedHat 9, 
I get exactly the same error when loading on a freshly sanitised system 
with all traces of previous asterisk installations removed.


HOWEVER, if you add a noload => pbx_wilcalu.so in modules.conf you can 
make the error go away. (but this is probably a bad thing since I don't 
know what that module does!)


But unfortunately, for me at least, I then end up with errors about:

app_capiCD.so
app_capiHOLD.so
app_capiRETRIEVE.so
app_capiECT.so
and
app_capiMCID.so


For example:

[app_capiCD.so]Nov 12 20:24:55 WARNING[19197]: loader.c:325 
__load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined 
symbol: ast_capi_MessageNumber
Nov 12 20:24:55 WARNING[19197]: loader.c:554 load_modules: Loading 
module app_capiCD.so failed!


# Ouch ... error while writing audio data: : Broken pipe

No matter which of the modules you comment out above, the same thing 
happens -- the error is always about app_capi_MessageNumber


Armin (or anybody) -- have I missed something out/done something wrong, 
or is it a compatibility issue between chan_capi-cm 0.6 and Asterisk 1.2 
RC2?



Faris.

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AW: [Asterisk-Users] chan_capi-cm-0.6 can't be loaded with latest asterisk version from cvs

2005-11-05 Thread mbodbg

Thanks Armin, this version is working, but I still have an undefined symbol
in another module:

...

[pbx_wilcalu.so]Nov  5 18:51:12 WARNING[11348]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol:
ast_pthread_create
Nov  5 18:51:12 WARNING[11348]: loader.c:554 load_modules: Loading module
pbx_wilcalu.so failed!

...

Can you also help me on that issue?

Thanks and Regards

Markus



-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Gesendet: Samstag, 5. November 2005 18:25
An: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Betreff: Re: [Asterisk-Users] chan_capi-cm-0.6 can't be loaded with latest
asterisk version from cvs

Please try chan_capi-cm CVS HEAD on sourceforge.net

Armin


On Sat, 5 Nov 2005 [EMAIL PROTECTED] wrote:
> Hello all,
> 
> I've been using chan_capi-cm-0.6 as CAPI channel driver, the driver was
> working fine until I've reinstalled asterisk last week. I retrieved the
> latest asterisk version from cvs and then build and installed it.
> 
> When rebuilding the capi channel driver with the latest asterisk headers I
> receive the following warning:
> 
> ... 
> 
> chan_capi.c:4014: Warning implicit declaration of function >>
> use_ast_mutex_init_intstead_of_pthread__mutex_init <<
> 
> ...
> 
> If I start asterisk, it fails with the following error:
> 
> [chan_capi.so]Nov  5 17:16:51 WARNING[2830]: loader.c:325 __load_resource:
> /usr/lib/asterisk/modules/chan_capi.so: undefined symbol:
> use_ast_mutex_init_instead_of_pthread_mutex_init
> Nov  5 17:16:51 WARNING[2830]: loader.c:554 load_modules: Loading module
> chan_capi.so failed!
> 
> Any help would be appreciated!
> 
> Markus
> 
> 
> 
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