[asterisk-users] CAPI didn't get a frame | avoiding initial deadlock | multiple instances of Asterisk

2007-12-10 Thread Marco Mouta
Hi guys,

First of all, I know that this server must be upgraded asap, I'm just
wondering if anyone of you has already faced this problem and , if so, would
the upgrade solve my problems...

CAPI version 0.6
Asterisk 1.2.5
AGI scripts are being used

Main problems:

-Dropped Calls

- ps aux | grep asterisk shows that asterisk (that is started with
safe_asterisk) is generating multiple instances of asterisk by it's self.

This may be caused by AGI scripts  http://bugs.digium.com/view.php?id=8086?

Extracted from  /var/log/asterisk/full :

Dec  6 10:11:07 DEBUG[5858] channel.c: Didn't get a frame from channel:
CAPI/ISDN1/141-721
Dec  6 10:11:07 DEBUG[5858] channel.c: Bridge stops bridging channels
CAPI/ISDN1/141-721 and SIP/741411-57bf
Dec  6 10:11:07 DEBUG[5858] chan_sip.c: update_call_counter(741411) -
decrement call limit counter
Dec  6 10:11:07 VERBOSE[5858] logger.c:   == Spawn extension (from-trunk,
141, 111) exited non-zero on 'CAPI/ISDN1/141-721'
Dec  6 10:11:07 VERBOSE[5858] logger.c: -- Executing
NoOp(CAPI/ISDN1/141-721, from-trunk - h - 112 - 112) in new stack
Dec  6 10:11:07 VERBOSE[5858] logger.c: -- Executing
AGI(CAPI/ISDN1/141-721, set_callerid.agi) in new stack
Dec  6 10:11:08 DEBUG[5858] res_agi.c: CAPI/ISDN1/141-721 hungup
Dec  6 10:11:08 VERBOSE[5858] logger.c:   == Spawn extension (from-trunk, h,
2) exited non-zero on 'CAPI/ISDN1/141-721'
Dec  6 10:11:08 DEBUG[2692] channel.c: Avoiding initial deadlock for
'CAPI/ISDN1/141-721'
Dec  6 10:11:08 DEBUG[2692] channel.c: Avoiding initial deadlock for
'CAPI/ISDN1/141-721'

Dec  6 10:13:02 DEBUG[5878] channel.c: Didn't get a frame from channel:
SIP/741411-f7c1
Dec  6 10:13:02 DEBUG[5878] channel.c: Bridge stops bridging channels
CAPI/ISDN1/141-722 and SIP/741411-f7c1

Dec  6 10:14:15 DEBUG[5902] channel.c: Didn't get a frame from channel:
CAPI/ISDN1/141-723
Dec  6 10:14:15 DEBUG[5902] channel.c: Bridge stops bridging channels
CAPI/ISDN1/141-723 and SIP/741411-4d0f
Dec  6 10:14:34 DEBUG[2692] channel.c: Avoiding initial deadlock for
'CAPI/ISDN1/141-725'
Dec  6 10:14:34 DEBUG[2692] channel.c: Avoiding initial deadlock for
'CAPI/ISDN1/141-725'

Dec  6 10:14:54 VERBOSE[5941] logger.c: -- SIP/741411-26c2 answered
CAPI/ISDN1/141-726
Dec  6 10:15:15 DEBUG[5941] channel.c: Didn't get a frame from channel:
SIP/741411-26c2
Dec  6 10:15:15 DEBUG[5941] channel.c: Bridge stops bridging channels
CAPI/ISDN1/141-726 and SIP/741411-26c2

Dec  6 10:45:29 DEBUG[6235] channel.c: Didn't get a frame from channel:
SIP/741411-9981
Dec  6 10:45:29 DEBUG[6235] channel.c: Bridge stops bridging channels
CAPI/ISDN1/132-72d and SIP/741411-9981


I definitely must go through CAPI 1.0 and asterisk 1.2.25, but would be
interesting to learn from this problem before upgrading, that's why I'm
posting on the list.

Could it be possible that the multiple instances of Asterisk that are
started due to the bug ( http://bugs.digium.com/view.php?id=8086, solved in
asterisk 1.2.13) are causing all the troubles, making this multiple
instances try to access same asterisk channel (leading us to Avoiding
deadlock messages) ?

I mean applying the patch might solve the problems instead off all system
upgrade?

Best regards,
Marco Mouta
-- 
Esta mensagem (incluindo quaisquer anexos) pode conter informação
confidencial para uso exclusivo do destinatário. Se não for o destinatário
pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu
esta mensagem por engano, por favor informe o emissor e elimine-a
imediatamente. Obrigado.

This e-mail message is intended only for individual(s) to whom it is
addressed and may contain information that is privileged, confidential,
proprietary, or otherwise exempt from disclosure under applicable law. If
you believe you have received this message in error, please advise the
sender by return e-mail and delete it from your mailbox. Thank you.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] capi installation problem

2007-03-07 Thread Giedrius Augys

Hello,
I have problem with capi, I can't install it. I have putted all info what I
did and what I get :). I think you can suggest me how to solve this
problem.and thanking you in anticipation. I have ISDN Frtiz!Card PCI
v2.1and I want  to install it to my ubuntu box (kernel:
2.6.17-10-server). Using command lspci -vv , I can see that kernel finds
this card:
*02:0d.0 Network controller: AVM Audiovisuelles MKTG  Computer System GmbH
Fritz!PCI v2.0 ISDN (rev 02)
   Subsystem: AVM Audiovisuelles MKTG  Computer System GmbH Fritz!PCI
v2.0 ISDN
   Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr-
Stepping- SERR+ FastB2B-
   Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort-
TAbort- MAbort- SERR- PERR-
   Interrupt: pin A routed to IRQ 201
   Region 0: Memory at ff9fec00 (32-bit, non-prefetchable) [size=32]
   Region 1: I/O ports at df80 [size=32]
   Capabilities: [40] Power Management version 2
   Flags: PMEClk- DSI- D1- D2+ AuxCurrent=55mA
PME(D0-,D1-,D2+,D3hot+,D3cold+)
   Status: D0 PME-Enable- DSel=0 DScale=0 PME-*



 I found tutorial in http://www.asteriskguru.com/tutorials/avm_b1.html .I
have installed these packages:
*ii  capiutils  3.9.20060704-1  Utilities for
CAPI-capable ISDN cards
ii  libcapi20-33.9.20060704-1  libraries for
CAPI support
ii  libcapi20-dev  3.9.20060704-1  libraries for
CAPI support
ii  avm-fritz-firmware-2.6.17-10   3.11+2.6.17.7-10.1  Firmware for AVM
Fritz! ISDN hardware*

and downloaded firmaware from *
ftp://ftp.in-berlin.de/pub/capi4linux/firmware/b1/ .*My capi.conf is:
*b1pci   /usr/share/isdn/b1.t4   DSS1-   -
-   -   P2P*

Then I exec command capiinit start, I have noting on output, but it load
modules:
[EMAIL PROTECTED]:~# lsmod
Module  Size  Used by
b1pci  10624  0
b1dma  17412  1 b1pci
b1 25856  2 b1pci,b1dma
capi   19392  0
kernelcapi 49664  4 b1pci,b1dma,b1,capi
capifs  7176  2 capi
ipv6  271136  12
lp 12964  0
mISDN_l2   44288  0
mISDN_l1   13192  0
avmfritz   25740  0
mISDN_isac 17280  1 avmfritz
mISDN_core 75648  4 mISDN_l2,mISDN_l1,avmfritz,mISDN_isac*

But when I execute command cappinfo, I get :
[EMAIL PROTECTED]:~# capiinfo
capi not installed - No such device or address (6).
*
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] capi installation problem

2007-03-07 Thread Armin Schindler
I think you are mixing something here. The FritzCard is not a B1, so you 
don't need the b1 modules, the firmware and the /etc/capi.conf.

You can either use the FritzCard driver (binary modules from AVM), or you 
use mISDN (which is also already loaded according to your lsmod).

When using mISDN, you can either use the mISDN-CAPI to really provide a CAPI
interface, or just don't use CAPI and use chan_misdn instead.

Armin

On Wed, 7 Mar 2007, Giedrius Augys wrote:
 Hello,
 I have problem with capi, I can't install it. I have putted all info what I
 did and what I get :). I think you can suggest me how to solve this
 problem.and thanking you in anticipation. I have ISDN Frtiz!Card PCI
 v2.1and I want  to install it to my ubuntu box (kernel:
 2.6.17-10-server). Using command lspci -vv , I can see that kernel finds
 this card:
 *02:0d.0 Network controller: AVM Audiovisuelles MKTG  Computer System GmbH
 Fritz!PCI v2.0 ISDN (rev 02)
Subsystem: AVM Audiovisuelles MKTG  Computer System GmbH Fritz!PCI
 v2.0 ISDN
Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr-
 Stepping- SERR+ FastB2B-
Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort-
 TAbort- MAbort- SERR- PERR-
 Interrupt: pin A routed to IRQ 201
 Region 0: Memory at ff9fec00 (32-bit, non-prefetchable) [size=32]
 Region 1: I/O ports at df80 [size=32]
 Capabilities: [40] Power Management version 2
Flags: PMEClk- DSI- D1- D2+ AuxCurrent=55mA
 PME(D0-,D1-,D2+,D3hot+,D3cold+)
 Status: D0 PME-Enable- DSel=0 DScale=0 PME-*
 
 
 
  I found tutorial in http://www.asteriskguru.com/tutorials/avm_b1.html .I
 have installed these packages:
 *ii  capiutils  3.9.20060704-1  Utilities for
 CAPI-capable ISDN cards
 ii  libcapi20-33.9.20060704-1  libraries for
 CAPI support
 ii  libcapi20-dev  3.9.20060704-1  libraries for
 CAPI support
 ii  avm-fritz-firmware-2.6.17-10   3.11+2.6.17.7-10.1  Firmware for AVM
 Fritz! ISDN hardware*
 
 and downloaded firmaware from *
 ftp://ftp.in-berlin.de/pub/capi4linux/firmware/b1/ .*My capi.conf is:
 *b1pci   /usr/share/isdn/b1.t4   DSS1-   -
 -   -   P2P*
 
 Then I exec command capiinit start, I have noting on output, but it load
 modules:
 [EMAIL PROTECTED]:~# lsmod
 Module  Size  Used by
 b1pci  10624  0
 b1dma  17412  1 b1pci
 b1 25856  2 b1pci,b1dma
 capi   19392  0
 kernelcapi 49664  4 b1pci,b1dma,b1,capi
 capifs  7176  2 capi
 ipv6  271136  12
 lp 12964  0
 mISDN_l2   44288  0
 mISDN_l1   13192  0
 avmfritz   25740  0
 mISDN_isac 17280  1 avmfritz
 mISDN_core 75648  4 mISDN_l2,mISDN_l1,avmfritz,mISDN_isac*
 
 But when I execute command cappinfo, I get :
 [EMAIL PROTECTED]:~# capiinfo
 capi not installed - No such device or address (6).
 *
 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] CAPI module issue

2006-12-03 Thread Esteban Guana-Jarrin

Hi List,

I am experiencing an issue with a server running asterisk; I installed an 
AVM FRITZ card and configured it to work with the capi module.


Once everything is installed the card works perfect; the issue is that every 
time I reboot the machine I have to re install the capi4k-utils before I can 
load asterisk otherwise the capi module does not loadup.


After boot up when I try capiinfo I get the following error message,

capi not installed - No such file or directory (2)

Once re-installed I get the following,

[EMAIL PROTECTED] ~]# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.11-02  (49.18)
Serial Number: 101
BChannels: 2

Furhter details follow:
Problem: capi4k-utils (capilib_new_ncci) does not load automatically after 
boot up

AVM FRTIZ card latest driver (BRI)
capi module: chan_capi-cm-0.6.5
capi4k-utils
Linux Distro: Fedora core 3
Kernel: 2.6.12-2.3.legacy_FC3 (capi supported by kernel)

Has any one experienced this before? Can anyone please provide with some 
ideas to overcome this issue?


Thanks in advance

Paul

_
Advertisement: Meet Sexy Singles Today @ Lavalife - Click here  
http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Flavalife9%2Eninemsn%2Ecom%2Eau%2Fclickthru%2Fclickthru%2Eact%3Fid%3Dninemsn%26context%3Dan99%26locale%3Den%5FAU%26a%3D23769_t=754951090_r=endtext_lavalife_dec_meet_m=EXT


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] CAPI module issue

2006-11-30 Thread Esteban Guana-Jarrin

Hi List,

I am experiencing an issue in a server that I have installed asterisk; 
configured an AVM FRITZ card to work with the capi module.


Once istalled the card works perfect; however every time I reboot the 
machine I found that I have to re install the capi4k-utils before I can load 
asterisk otherwise the capi module will not loadup.


Can anyone direct me in the right direction in order to find out why 
capi4k-utils (capilib_new_ncci) do not load automatically after boot up?


Details follow:
AVM FRTIZ card latest driver (BRI)
capi module: chan_capi-cm-0.6.5
capi4k-utils
Linux Distro: Fedora core 3
Kernel: 2.6.12-2.3.legacy_FC3 (capi supported by kernel)

Thanks in advance

Paul

_
Advertisement: It's simple! Sell your car for just $20 at carsales.com.au  
http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Fsecure%2Dau%2Eimrworldwide%2Ecom%2Fcgi%2Dbin%2Fa%2Fci%5F450304%2Fet%5F2%2Fcg%5F801577%2Fpi%5F1005244%2Fai%5F838588_t=757768878_r=endtext_simple_m=EXT


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] CAPI channel not available but nobody is using the system

2006-10-18 Thread Armin Schindler
On Tue, 17 Oct 2006, Tim Sharp wrote:
 I have 23 CAPI channels defined and normally multiple channels are in use 
 during the day for outbound calling.  The problem is that every 3 or 4 
 months one of the channels becomes unavailable and then no calls can come 
 in or go out on any of these channels.  CAPI INFO shows Contr1: 23 B 
 channels total, 22 B channels free.  To fix the problem I reboot the 
 asterisk server.  First, is there a better way to reset the channels than 
 rebooting?

It depends where the problem really has its origin.
If just asterisk (chan-capi) has a wrong channel count, it would be enough
to unload chan-capi. Maybe asterisk itself need to be restarted.
But if the real problem comes from the CAPI/ISDN driver, you need to reload
these drivers. 

Which version of asterisk/chan-capi do you use? What ISDN hardware / CAPI 
driver do you use?

 Second, is there a way to bypass the unavailable channel in the dialplan?

No.

 Third, what is causing the problem and can I prevent it? 

chan-capi counts the active channels when the CONNECT/DISCONNECT message
of b-channels are indicated. If one of these messages are missing (it's a 
bug in the CAPI driver if that happens) the count is wrong.


Armin


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [asterisk-users] CAPI channel not available but nobody is usingthe system

2006-10-18 Thread Tim Sharp
Armin,
I am running 1.2.7.1 with an Eicon T1 board version 2 on Debian 2.4
I don't know the details on chan-capi / CAPI drivers.  We did the install April 
of this year.
How can I tell what I have?
Thank you for your time.
Tim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Armin
Schindler
Sent: Wednesday, October 18, 2006 3:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CAPI channel not available but nobody is
usingthe system


On Tue, 17 Oct 2006, Tim Sharp wrote:
 I have 23 CAPI channels defined and normally multiple channels are in use 
 during the day for outbound calling.  The problem is that every 3 or 4 
 months one of the channels becomes unavailable and then no calls can come 
 in or go out on any of these channels.  CAPI INFO shows Contr1: 23 B 
 channels total, 22 B channels free.  To fix the problem I reboot the 
 asterisk server.  First, is there a better way to reset the channels than 
 rebooting?

It depends where the problem really has its origin.
If just asterisk (chan-capi) has a wrong channel count, it would be enough
to unload chan-capi. Maybe asterisk itself need to be restarted.
But if the real problem comes from the CAPI/ISDN driver, you need to reload
these drivers. 

Which version of asterisk/chan-capi do you use? What ISDN hardware / CAPI 
driver do you use?

 Second, is there a way to bypass the unavailable channel in the dialplan?

No.

 Third, what is causing the problem and can I prevent it? 

chan-capi counts the active channels when the CONNECT/DISCONNECT message
of b-channels are indicated. If one of these messages are missing (it's a 
bug in the CAPI driver if that happens) the count is wrong.


Armin


___
--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
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [asterisk-users] CAPI channel not available but nobody is usingthe system

2006-10-18 Thread Armin Schindler
On Wed, 18 Oct 2006, Tim Sharp wrote:
 Armin,
 I am running 1.2.7.1 with an Eicon T1 board version 2 on Debian 2.4
 I don't know the details on chan-capi / CAPI drivers.  We did the install 
 April of this year.
 How can I tell what I have?

The divas driver version can be found in the syslog messages when the driver 
is loaded.
I recommend to use the new V3 driver (ftp.melware.net).

When you start asterisk (with verbosity 5) you can see the chan-capi 
messages including its version. 
It's an too old version if it is from April, please update, same ftp-server.

Armin

 Thank you for your time.
 Tim
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Armin
 Schindler
 Sent: Wednesday, October 18, 2006 3:24 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] CAPI channel not available but nobody is
 usingthe system
 
 
 On Tue, 17 Oct 2006, Tim Sharp wrote:
  I have 23 CAPI channels defined and normally multiple channels are in use 
  during the day for outbound calling.  The problem is that every 3 or 4 
  months one of the channels becomes unavailable and then no calls can come 
  in or go out on any of these channels.  CAPI INFO shows Contr1: 23 B 
  channels total, 22 B channels free.  To fix the problem I reboot the 
  asterisk server.  First, is there a better way to reset the channels than 
  rebooting?
 
 It depends where the problem really has its origin.
 If just asterisk (chan-capi) has a wrong channel count, it would be enough
 to unload chan-capi. Maybe asterisk itself need to be restarted.
 But if the real problem comes from the CAPI/ISDN driver, you need to reload
 these drivers. 
 
 Which version of asterisk/chan-capi do you use? What ISDN hardware / CAPI 
 driver do you use?
 
  Second, is there a way to bypass the unavailable channel in the dialplan?
 
 No.
 
  Third, what is causing the problem and can I prevent it? 
 
 chan-capi counts the active channels when the CONNECT/DISCONNECT message
 of b-channels are indicated. If one of these messages are missing (it's a 
 bug in the CAPI driver if that happens) the count is wrong.
 
 
 Armin
 
 
 ___
 --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
 ___
 --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
 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] CAPI channel not available but nobody is using the system

2006-10-17 Thread Tim Sharp
I have 23 CAPI channels defined and normally multiple channels are in use 
during the day for outbound calling.  The problem is that every 3 or 4 months 
one of the channels becomes unavailable and then no calls can come in or go out 
on any of these channels.  CAPI INFO shows Contr1: 23 B channels total, 22 B 
channels free.  To fix the problem I reboot the asterisk server.  First, is 
there a better way to reset the channels than rebooting?  Second, is there a 
way to bypass the unavailable channel in the dialplan?  Third, what is causing 
the problem and can I prevent it?
Thank you in advance.
Tim
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] capi (divas4linux) bearer setting

2006-08-16 Thread Farkas Levente
hi,
we've got a Diva Server BRI-2M PCI, SN:3485 card in our asterisk server.
we use the latest divas4linux-melware-3.0.g-106.628.1-1 driver for it.
the card is connected to a Bosch Integral33 PBX. the two system
connected with an S0 line in order the two pbx be able to call
eachother. when we call from the bosch to asterisk everything is working
properly. but when we call from the a x-ten soft phone client through
asterisk to the bosch the it's not working. which means the asterisk
pass the call to the bosch, bosch receive but don't ring the given
number. after we debug the capi layer with bosch experts from bosch we
found the while the bosch call asterisk it request SPEECH time bearer,
but when asterisk call bosch it set bearer to MULTIUSE. i found it in
./divactrl/common/dbg_tapi.c LINE_BEARER_MODE__SPEECH,
LINE_BEARER_MODE__MULTIUSE. so probably the problem is thet we (x-ten,
asterisk, a divas4linux do not set the bearer to proper value. is this
the real reason? how can i set the bearer to speech in divas4linux or in
capi or in asterisk's capi or ...?
thank you for your help in advance.
yours.

-- 
  Levente   Si vis pacem para bellum!
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] capi drivers for suse-10.1

2006-06-04 Thread Hans Witvliet
For those with an Fritz!board, have a look at:
http://www.fltronic.de/~olly/avm/

-- 
pgp-id: 926EBB12
pgp-fingerprint: BE97 1CBF FAC4 236C 4A73  F76E EDFC D032 926E BB12
Registered linux user: 75761 (http://counter.li.org)
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] CAPI Installation Eicon Diva Server

2006-04-18 Thread nkohl
Hi Avi 

This is great - the problem was how I configured my trunk so this part
of your v. good wiki page was my solution:

-


Maximum channels: num of ports * 2
I have 2 ISDN lines active, so I have 4 maximum channels. If you have
all 4 ports running, you have 8 maximum channels. Each ISDN line has 2
channels.

Custom dial string: CAPI/g1/$OUTNUM$/b

Alternatively, you could configure a trunk per port by using:

CAPI/Contr1/$OUTNUM$/b

You need to set 2 maximum channels for each port.

-

Too bad the documentation is a little sketchy on this stuff... 

Cheers,
Nick

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller
Sent: 12 April 2006 22:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CAPI Installation Eicon Diva Server

[EMAIL PROTECTED] wrote:
 Asterisk says it has 30 capi channels available, but my mistake may be

 in configuring the trunks...

When I was debugging my Eicon Diva 4-BRI board, I found it useful to
play with extensions_custom.conf (in AMP) just to ensure I got the
Custom Dial String absolutely correct. According to the latest
chan_capi-cm, the Dial String should be:

CAPI/id/number/options

Where:

id = Contr1 or g1 (Controller or Group ID) number = Phone number
options = Things like B or b for Early B3 and other things. I have 'b'

in my options, but I do admit that I have no idea what early B3 is. :)

Hope that helps in some way,
Avi

P.S. I wrote a quick config page for the 4-BRI for freePBX here: 
http://aussievoip.com/wiki/index.php?page=freePBX-EiconDiva

It might have a few things to consider as well.

--
National Manager - Special Projects

 Melbourne / Sydney / Canberra / Hobart / London /
   2/340 Gore StreetT: +61 (0) 3 9486 0411
   Fitzroy, VIC F: +61 (0) 3 9486 0611
   3065 W: http://www.squiz.net/

. Open Source  - Own it  -  Squiz.net ./
___
--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



**
Any information in this communication which is confidential must not
be disclosed to others without our consent. Such consent is not required
where the information is publicly available and intended for onward
distribution. If the information is confidential and if you are not the
intended recipient, you are not authorised to and must not disclose,
copy, distribute, or retain this message or any part of it. You are
requested to return this message to the sender immediately.

Due to the electronic nature of e-mail, there is a risk that the
information contained in this message has been modified. 
Consequently Man Investments can accept no responsibility or
liability as to the completeness or accuracy of the information.

Visit us at: www.maninvestments.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


Re: [Asterisk-Users] CAPI Installation Eicon Diva Server

2006-04-17 Thread Armin Schindler
On Wed, 12 Apr 2006 [EMAIL PROTECTED] wrote:
 Hi 
  
 I've got a dell 2550 with an Eicon Diva server PRI card plugged into it.
 I can call out using the acopy2 test utility.
  
 I'm having trouble with asterisk making calls however... my capi.conf
 and modules.conf looks correct by the wiki instructions - does anyone
 have any advice on where to look ? I can attach conf files etc. if
 needed.
  
 Asterisk says it has 30 capi channels available, but my mistake may be
 in configuring the trunks... 

The configuration is as easy as with BRI lines. Can you provide more (like 
your confs and verbose/debug output)?

Armin

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI Installation Eicon Diva Server

2006-04-17 Thread Avi Miller

Armin Schindler wrote:
The configuration is as easy as with BRI lines. Can you provide more (like 
your confs and verbose/debug output)?


Also (this isn't directed at you Armin, but I found your email to reply 
off of to maintain the threading), I created a Wiki page over at the 
freePBX documentation site, explaining how to configure an Eicon Server 
4-BRI for freePBX. It may have some tips for you:


http://aussievoip.com/wiki/index.php?page=freePBX-EiconDiva

Feel free to add/remove information. Its a Wiki after all. :)

cYa,
Avi

--
National Manager - Special Projects

 Melbourne / Sydney / Canberra / Hobart / London /
  2/340 Gore StreetT: +61 (0) 3 9486 0411
  Fitzroy, VIC F: +61 (0) 3 9486 0611
  3065 W: http://www.squiz.net/

. Open Source  - Own it  -  Squiz.net ./
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI Installation Eicon Diva Server

2006-04-12 Thread nkohl



Hi 


I've got a 
dell 2550 with an Eicon Diva server PRI card plugged into it. I can call out 
using the acopy2 test utility.

I'm having 
trouble with asterisk making calls however... my capi.conf and modules.conf looks correct by the wiki instructions - does anyone have any advice on where to 
look ? I can attach conf files etc. if needed.

Asterisk 
says it has 30 capi channels available, but my mistake may be in configuring the 
trunks... 

Nick

**
Any information in this communication which is confidential must not
be disclosed to others without our consent. Such consent is not required
where the information is publicly available and intended for onward
distribution. If the information is confidential and if you are not the
intended recipient, you are not authorised to and must not disclose,
copy, distribute, or retain this message or any part of it. You are
requested to return this message to the sender immediately.

Due to the electronic nature of e-mail, there is a risk that the
information contained in this message has been modified. 
Consequently Man Investments can accept no responsibility or
liability as to the completeness or accuracy of the information.

Visit us at: www.maninvestments.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


Re: [Asterisk-Users] CAPI Installation Eicon Diva Server

2006-04-12 Thread Avi Miller

[EMAIL PROTECTED] wrote:
Asterisk says it has 30 capi channels available, but my mistake may be 
in configuring the trunks...


When I was debugging my Eicon Diva 4-BRI board, I found it useful to 
play with extensions_custom.conf (in AMP) just to ensure I got the 
Custom Dial String absolutely correct. According to the latest 
chan_capi-cm, the Dial String should be:


CAPI/id/number/options

Where:

id = Contr1 or g1 (Controller or Group ID)
number = Phone number
options = Things like B or b for Early B3 and other things. I have 'b' 
in my options, but I do admit that I have no idea what early B3 is. :)


Hope that helps in some way,
Avi

P.S. I wrote a quick config page for the 4-BRI for freePBX here: 
http://aussievoip.com/wiki/index.php?page=freePBX-EiconDiva


It might have a few things to consider as well.

--
National Manager - Special Projects

 Melbourne / Sydney / Canberra / Hobart / London /
  2/340 Gore StreetT: +61 (0) 3 9486 0411
  Fitzroy, VIC F: +61 (0) 3 9486 0611
  3065 W: http://www.squiz.net/

. Open Source  - Own it  -  Squiz.net ./
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI crash/lockups?

2006-01-24 Thread Michael J. Tubby B.Sc \(Hons\) G8TIC



List (and Armin)

My set up is as follows:

 RedHat Fedora Core #3 
box
 AVM C4 card
 Asterisk 1.2.1
 chan_capi-cm-0.6.1 (with patches 
from Armin)
 2 x ISDN2e line (4 channels) 
bonded in P2P mode - British Telecom

I have, before today, had two occasions where the 
CAPI sub-system just "stopped", ie. I could not place or receive calls - 
attempting to place calls resulted in Reorder at the SIP phone - calling our 
switchboard number resulted in 5 seconds silence and then number 
un-obtainable.

Stopping and re-starting Asterisk did not fix the 
problem - instead I had to power down the server and re-start it - presumably to 
re-start the C4 card and its firmware?

On the previous occasions I could plug my older 
Panasonic KXTD816 exchange in to the ISDN2e lines and the ISDN would come up and 
calls could be made/received.

Today we had a more significant 
failure:

a) the console had the following printk() messages 
on it:

 kcapi: msgid: 42885 ncci 0x10a01 
not on queue
 capilib_free_ncci: ncci 0x10101 
not found
 kcapi: msgid:55642 ncci 
0x10201 not on queue

b) enabling "capi debug" and attempting to place an 
outgoing call failed with "reason 44 - channel not available"

c) plugging in the old Panasonic exchange - the 
ISDN did not work - BT announcement "alert Sorry there is a temportary 
fault"

I had to get BT to clear the fault and reboot the 
asterisk system to get back on line.


Advice sought...



Regards


Mike


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI unable to handle busy()

2006-01-03 Thread Karsten Wemheuer
Hello Armin,

On Mo, 2 Jan 2006 Armin Schindler wrote:
 I don't think it is necessary to exclude it. Just build chan_capi-cm and 
 overwrite chan_capi.so as well as remove the app_capi* modules from your 
 installation.
 
 Armin

Many thanks, it is working.

Karsten

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI unable to handle busy()

2006-01-02 Thread Karsten Wemheuer
Hello,

first of all, I say Happy New Year to this list!

While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which includes
chan_capi 0.4.0-PRE1), I ran into the following problem.

I want to signal busy to an incoming call, but that doesn't work.

The dialplan looks like this:
exten = 22715292,1,Busy
(The extension is ok and works fine, if I use other applications like
Dial)

The result is:
   -- creating pipe for PLCI=0x101 msn = 22715292
sent ALERT_REQ PLCI = 0x101
-- Executing Busy(CAPI/contr1/22715292-13, ) in new stack
-- started pbx on channel (callgroup=0)!

The caller hears still ringing signal.

If I replace Busy with Busy(2), the following happens:
-- creating pipe for PLCI=0x101 msn = 22715292
sent ALERT_REQ PLCI = 0x101
-- Executing Busy(CAPI/contr1/22715292-14, 2) in new stack
-- started pbx on channel (callgroup=0)!
  == Spawn extension (incoming, 22715292, 1) exited non-zero on
'CAPI/contr1/22715292-14'
-- CAPI Hangingup
-- removed pipe for PLCI = 0x101
But again, the calling site gets no busy-signalling.

If I use hangup(17) instead of busy() (which should be the same as 17 is
the value for the busy condition), I get the following result:
   -- creating pipe for PLCI=0x101 msn = 22715292
sent ALERT_REQ PLCI = 0x101
-- Executing Hangup(CAPI/contr1/22715292-15, 17) in new stack
  == Spawn extension (incoming, 22715292, 1) exited non-zero on
'CAPI/contr1/22715292-15'
-- CAPI Hangingup
sent CONNECT_RESP for PLCI = 0x101
-- removed pipe for PLCI = 0x101
-- started pbx on channel (callgroup=0)!
Jan  2 14:00:36 ERROR[1143]: chan_capi.c:1237 pipe_frame: wrote -1 bytes
instead of 48

The calling site will see a normal call clearing.

Hardware is a FritzPCI! (AVM).

If I do the same things with a HFC-based card and chan_zap, both version
(busy() and hangup(17)) are working fine.

Any helping hints are welcome!

Thanks!

Karsten

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI unable to handle busy()

2006-01-02 Thread Armin Schindler
On Mon, 2 Jan 2006, Karsten Wemheuer wrote:
 Hello,
 
 first of all, I say Happy New Year to this list!
 
 While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which includes
 chan_capi 0.4.0-PRE1), I ran into the following problem.
 
 I want to signal busy to an incoming call, but that doesn't work.
 
 The dialplan looks like this:
   exten = 22715292,1,Busy
 (The extension is ok and works fine, if I use other applications like
 Dial)

chan_capi from junghanns/bristuff does not support that.
I suggest using the new chan_capi-cm-0.6.2

Armin
 
 The result is:
-- creating pipe for PLCI=0x101 msn = 22715292
 sent ALERT_REQ PLCI = 0x101
 -- Executing Busy(CAPI/contr1/22715292-13, ) in new stack
 -- started pbx on channel (callgroup=0)!
 
 The caller hears still ringing signal.
 
 If I replace Busy with Busy(2), the following happens:
 -- creating pipe for PLCI=0x101 msn = 22715292
 sent ALERT_REQ PLCI = 0x101
 -- Executing Busy(CAPI/contr1/22715292-14, 2) in new stack
 -- started pbx on channel (callgroup=0)!
   == Spawn extension (incoming, 22715292, 1) exited non-zero on
 'CAPI/contr1/22715292-14'
 -- CAPI Hangingup
 -- removed pipe for PLCI = 0x101
 But again, the calling site gets no busy-signalling.
 
 If I use hangup(17) instead of busy() (which should be the same as 17 is
 the value for the busy condition), I get the following result:
-- creating pipe for PLCI=0x101 msn = 22715292
 sent ALERT_REQ PLCI = 0x101
 -- Executing Hangup(CAPI/contr1/22715292-15, 17) in new stack
   == Spawn extension (incoming, 22715292, 1) exited non-zero on
 'CAPI/contr1/22715292-15'
 -- CAPI Hangingup
 sent CONNECT_RESP for PLCI = 0x101
 -- removed pipe for PLCI = 0x101
 -- started pbx on channel (callgroup=0)!
 Jan  2 14:00:36 ERROR[1143]: chan_capi.c:1237 pipe_frame: wrote -1 bytes
 instead of 48
 
 The calling site will see a normal call clearing.
 
 Hardware is a FritzPCI! (AVM).
 
 If I do the same things with a HFC-based card and chan_zap, both version
 (busy() and hangup(17)) are working fine.
 
 Any helping hints are welcome!
 
 Thanks!
 
 Karsten
 
 ___
 --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
 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI unable to handle busy()

2006-01-02 Thread Karsten Wemheuer
Hello Armin,

On Mo, 02.01.2006 Armin Schindler wrote:
 On Mon, 2 Jan 2006, Karsten Wemheuer wrote:
  Hello,
  
  first of all, I say Happy New Year to this list!
  
  While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which includes
  chan_capi 0.4.0-PRE1), I ran into the following problem.
  
  I want to signal busy to an incoming call, but that doesn't work.
  
  The dialplan looks like this:
  exten = 22715292,1,Busy
  (The extension is ok and works fine, if I use other applications like
  Dial)
 
 chan_capi from junghanns/bristuff does not support that.
 I suggest using the new chan_capi-cm-0.6.2
 

thanks for the quick response. How can I implement bristuff-patch and
Your new chan_capi? I need a version with both, ZAP-HFC and CAPI. So the
question is, how can I exclude chan_capi from the bristuff-patches?

Any Ideas?

Thanks

Karsten

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI unable to handle busy()

2006-01-02 Thread Armin Schindler
On Mon, 2 Jan 2006, Karsten Wemheuer wrote:
 Hello Armin,
 
 On Mo, 02.01.2006 Armin Schindler wrote:
  On Mon, 2 Jan 2006, Karsten Wemheuer wrote:
   Hello,
   
   first of all, I say Happy New Year to this list!
   
   While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which includes
   chan_capi 0.4.0-PRE1), I ran into the following problem.
   
   I want to signal busy to an incoming call, but that doesn't work.
   
   The dialplan looks like this:
 exten = 22715292,1,Busy
   (The extension is ok and works fine, if I use other applications like
   Dial)
  
  chan_capi from junghanns/bristuff does not support that.
  I suggest using the new chan_capi-cm-0.6.2
  
 
 thanks for the quick response. How can I implement bristuff-patch and
 Your new chan_capi? I need a version with both, ZAP-HFC and CAPI. So the
 question is, how can I exclude chan_capi from the bristuff-patches?

I don't think it is necessary to exclude it. Just build chan_capi-cm and 
overwrite chan_capi.so as well as remove the app_capi* modules from your 
installation.

Armin

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI and *

2005-12-31 Thread Sascha Andres
Hi,
* Armin Schindler wrote on 24.12.2005 (13:18):
 I suggest you use chan_capi-cm from sourceforge.net instead of old 
 0.3.5/0.4.0. And when installing a new version, remove old files from 
 installation like app_capi*

done that. Now I got a bunch of other problems. I don't
think they're related to asterisk but to basic CAPI
configuration. It turns out that capi configuration can be a
nightmare :(

Sascha

-- 
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI and *

2005-12-31 Thread Armin Schindler
On Sat, 31 Dec 2005, Sascha Andres wrote:
 Hi,
 * Armin Schindler wrote on 24.12.2005 (13:18):
  I suggest you use chan_capi-cm from sourceforge.net instead of old 
  0.3.5/0.4.0. And when installing a new version, remove old files from 
  installation like app_capi*
 
 done that. Now I got a bunch of other problems. I don't
 think they're related to asterisk but to basic CAPI
 configuration. It turns out that capi configuration can be a
 nightmare :(

CAPI itself is very easy, but the card drivers are different...

What problems do you have ?

Armin

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI and *

2005-12-31 Thread Sascha Andres
Hi,
* Armin Schindler wrote on 31.12.2005 (12:47):
 CAPI itself is very easy, but the card drivers are different...
 
 What problems do you have ?

Every application (not only asterisk) complains that capi is
not loaded. I do have two choices for my isdn card: AVM B1
PCMCIA and Eicon Diva Mobile V90. I prefer the last on,
because I have two of them and want to connect a isdn phone
to it. I think the AVM card isn't capable running in nt
mode.

So far all modules are loaded and there doesn't seem to be an
error in /var/log/messages.

My loaded modules:

,[ module_list ]-
| divacapi  157188  0 
| divas  69324  0 
| divadidd   11584  2 divacapi,divas
| kernelcapi 44320  7 b1pci,b1dma,b1pcmcia,b1,divacapi,capidrv,capi
| b1pci   9472  0 
| b1dma  14980  1 b1pci
| b1pcmcia6528  0 
| b1 21632  3 b1pci,b1dma,b1pcmcia
| capidrv27572  0 
| isdn  121196  1 capidrv
| capi   16960  0 
| capifs  5768  2 capi
`

capiinfo shows the error 'capi not installed - No such
device or adress (6)'. A google search brought some tips but
they doesn't seem to be related to my problem. Most tips are
for passive pci,iso or usb cards.

When I shut down unloading capi complains about busy
kernelcapi - not sure how to track this down.

Kind regards and a happy new year,
Sascha

-- 
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI and *

2005-12-31 Thread Armin Schindler
On Sat, 31 Dec 2005, Sascha Andres wrote:
 Hi,
 * Armin Schindler wrote on 31.12.2005 (12:47):
  CAPI itself is very easy, but the card drivers are different...
  
  What problems do you have ?
 
 Every application (not only asterisk) complains that capi is
 not loaded. I do have two choices for my isdn card: AVM B1
 PCMCIA and Eicon Diva Mobile V90. I prefer the last on,
 because I have two of them and want to connect a isdn phone
 to it. I think the AVM card isn't capable running in nt
 mode.
 
 So far all modules are loaded and there doesn't seem to be an
 error in /var/log/messages.
 
 My loaded modules:
 
 ,[ module_list ]-
 | divacapi  157188  0 
 | divas  69324  0 
 | divadidd   11584  2 divacapi,divas
 | kernelcapi 44320  7 
 b1pci,b1dma,b1pcmcia,b1,divacapi,capidrv,capi
 | b1pci   9472  0 
 | b1dma  14980  1 b1pci
 | b1pcmcia6528  0 
 | b1 21632  3 b1pci,b1dma,b1pcmcia
 | capidrv27572  0 
 | isdn  121196  1 capidrv
 | capi   16960  0 
 | capifs  5768  2 capi
 `
 
 capiinfo shows the error 'capi not installed - No such
 device or adress (6)'. A google search brought some tips but
 they doesn't seem to be related to my problem. Most tips are
 for passive pci,iso or usb cards.
 
 When I shut down unloading capi complains about busy
 kernelcapi - not sure how to track this down.

The open-source diva driver (divas) does not support the Eicon Diva Mobile,
the Diva Server Cards are available only.
I don't know if the drivers from Eicon do support this card.
But anyway, I don't think this card is capable doing NT-mode.

The error 'capi not installed' just means, that there is no card/driver
registered which provides CAPI 2.0 interface.
For example the divas driver, just loading it (and divacapi) does not
provide a CAPI card. The card itself must be loaded with the firmware and 
started (divactrl load command).

Armin

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI and *

2005-12-31 Thread Sascha Andres
Hi,
* Armin Schindler wrote on 31.12.2005 (15:26):
 The open-source diva driver (divas) does not support the Eicon Diva Mobile,
 the Diva Server Cards are available only.
 I don't know if the drivers from Eicon do support this card.
 But anyway, I don't think this card is capable doing NT-mode.

The eicon driver itself doen't support it. So I can't use
this card :( I removed the modules from getting loaded.

At least the AVM card should be supported? If not, what card
(it should be active because I need to run my laptop on a
ISDN port that only active cards) should I go for?

Thanks,
Sascha

-- 
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI and *

2005-12-31 Thread Armin Schindler
On Sat, 31 Dec 2005, Sascha Andres wrote:
 Hi,
 * Armin Schindler wrote on 31.12.2005 (15:26):
  The open-source diva driver (divas) does not support the Eicon Diva Mobile,
  the Diva Server Cards are available only.
  I don't know if the drivers from Eicon do support this card.
  But anyway, I don't think this card is capable doing NT-mode.
 
 The eicon driver itself doen't support it. So I can't use
 this card :( I removed the modules from getting loaded.
 
 At least the AVM card should be supported? If not, what card
 (it should be active because I need to run my laptop on a
 ISDN port that only active cards) should I go for?

The AVM should work, but as far as I know not in NT-mode.

I don't have any knowledge about cards for laptops, sorry.

Armin

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI and *

2005-12-24 Thread Sascha Andres
Hi,

I got the newest asterisk (SVN-trunk-r7413) that compiled
fine without any errors or warnings. I got chan_capi 0.4
PRE1 and modified the sources together with a friend so 
ina way that no error or warning occurs.

When I try to load chan_capi the following error is printed
and asterisk quits:

,[ capi ]-
| [app_capiCD.so]Dec 24 12:57:16 WARNING[9060]: loader.c:325 __load_resource: 
/usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: 
ast_capi_MessageNumber
| Dec 24 12:57:16 WARNING[9060]: loader.c:554 load_modules: Loading module 
app_capiCD.so failed!
`

(Sorry for the long lines, I don't want to break the
messaged).

I'm not sure where to ask how to solve this, so I'm just
asking here.

Any help appreciated and have a nice christmas,
Sascha

-- 
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI and *

2005-12-24 Thread Armin Schindler
On Sat, 24 Dec 2005, Sascha Andres wrote:
 Hi,
 
 I got the newest asterisk (SVN-trunk-r7413) that compiled
 fine without any errors or warnings. I got chan_capi 0.4
 PRE1 and modified the sources together with a friend so 
 ina way that no error or warning occurs.
 
 When I try to load chan_capi the following error is printed
 and asterisk quits:
 
 ,[ capi ]-
 | [app_capiCD.so]Dec 24 12:57:16 WARNING[9060]: loader.c:325 __load_resource: 
 /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: 
 ast_capi_MessageNumber
 | Dec 24 12:57:16 WARNING[9060]: loader.c:554 load_modules: Loading module 
 app_capiCD.so failed!
 `
 
 (Sorry for the long lines, I don't want to break the
 messaged).
 
 I'm not sure where to ask how to solve this, so I'm just
 asking here.
 
 Any help appreciated and have a nice christmas,

I suggest you use chan_capi-cm from sourceforge.net instead of old 
0.3.5/0.4.0. And when installing a new version, remove old files from 
installation like app_capi*

Armin

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] capi incoming call timeout

2005-12-12 Thread Louis-David Mitterrand
Hello,

Using * 1.2.1 with chan_capi CVS on a Diva server I am mostly happy.
However when a phone redirects a call (user forward) and all ISDN
channels are busy, the call goes out through an IAX connection and it
takes a few seconds to get a ring state from the remote * server. This
makes the incoming call (on the Diva) timeout and the caller gets a
telco congestion tone. 

This can be solved by adding a fake ring (r) on the IAX connection
Dial() string, as the incoming call now gets a ringing state signaled to
it. 

Is there a way to increase the signaling timeout on the incoming call,
so that no fake ringing is required during the IAX call forward?

-- 
I had no wish to arrive, but I had to do my utmost, in order to
arrive. -- Samuel Beckett, The Unnamable
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Capi problem

2005-11-13 Thread Armin Schindler
Did you load the kernel module 'capi.o' as well? This is the module which 
provides the node /dev/capi20.

If you use mISDN, you don't need capiinit, which is for AVM drivers only.

Armin


On Sat, 12 Nov 2005, MBIT Technologies wrote:

 Hi Guys
 
  
 
 I'm having a problem getting CAPI to work on my Traverse NetJet card.
 
  
 
 CAPI is enabled in the kernel and I'm using the mISDN drivers with the
 NetJet patch. I cant seem to get astcapi to load
 
  
 
 Heres the output im getting
 
  
 
 Nov 12 21:18:36 VERBOSE[4011]:   == Registered application 'WaitMusicOnHold'
 
 Nov 12 21:18:36 VERBOSE[4011]:   == Registered application 'SetMusicOnHold'
 
 Nov 12 21:18:36 VERBOSE[4011]:  [chan_capi.so]Nov 12 21:18:36 VERBOSE[4011]:
 [chan_capi.so] = (Common ISDN API for Asterisk)
 
 Nov 12 21:18:36 VERBOSE[4011]:   == Parsing '/etc/asterisk/capi.conf': Nov
 12 21:18:36 VERBOSE[4011]:   == Parsing '/etc/asterisk/capi.conf': Found
 
 Nov 12 21:18:36 VERBOSE[4011]: -- ast_capi_pvt ISDN1 (*,from-pstn,0,2)
 (0,4,64)
 
 Nov 12 21:18:36 VERBOSE[4011]: -- ast_capi_pvt ISDN1 (*,from-pstn,0,2)
 (0,4,64)
 
 Nov 12 21:18:36 WARNING[4011]: CAPI not installed, CAPI disabled!
 
 Nov 12 21:18:36 WARNING[4011]: chan_capi.so: load_module failed, returning
 -1
 
 Nov 12 21:18:36 VERBOSE[4011]:   == Unregistered channel type 'CAPI'
 
 Nov 12 21:18:36 WARNING[4011]: Loading module chan_capi.so failed!
 
  
 
 I think it could be a udev problem so I put a file called 10-capi.rules in
 my udev directory with the following
 
  
 
 SYSFS(dev)=68:0,  NAME=capi20
 
 SYSFS(dev)=191:[0-9]*,NAME=capi/%n
 
  
 
 When I do a capiinit it says
 
  
 
 ERROR: cannot load module kernelcapi
 
  
 
 Does there need to be another entry for kernelcapi?
 
  
 
  
 
 Any help would be greatly appreciated.
 
  
 
 
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Capi problem

2005-11-12 Thread MBIT Technologies








Hi Guys



Im having a problem getting CAPI to work on my
Traverse NetJet card.



CAPI is enabled in the kernel and Im using the mISDN
drivers with the NetJet patch. I cant seem to get astcapi to load



Heres the output im getting



Nov 12 21:18:36 VERBOSE[4011]:
== Registered application 'WaitMusicOnHold'

Nov 12 21:18:36 VERBOSE[4011]:
== Registered application 'SetMusicOnHold'

Nov 12 21:18:36 VERBOSE[4011]:
[chan_capi.so]Nov 12 21:18:36
VERBOSE[4011]: [chan_capi.so] = (Common ISDN API for Asterisk)

Nov 12 21:18:36 VERBOSE[4011]:
== Parsing '/etc/asterisk/capi.conf': Nov 12 21:18:36
VERBOSE[4011]: == Parsing '/etc/asterisk/capi.conf': Found

Nov 12 21:18:36 VERBOSE[4011]:
-- ast_capi_pvt ISDN1 (*,from-pstn,0,2) (0,4,64)

Nov 12 21:18:36 VERBOSE[4011]:
-- ast_capi_pvt ISDN1 (*,from-pstn,0,2) (0,4,64)

Nov 12 21:18:36 WARNING[4011]:
CAPI not installed, CAPI disabled!

Nov 12 21:18:36 WARNING[4011]: chan_capi.so:
load_module failed, returning -1

Nov 12 21:18:36 VERBOSE[4011]:
== Unregistered channel type 'CAPI'

Nov 12 21:18:36 WARNING[4011]:
Loading module chan_capi.so failed!



I think it could be a udev problem so I put a file called
10-capi.rules in my udev directory with the following



SYSFS(dev)=68:0,
NAME=capi20

SYSFS(dev)=191:[0-9]*,
NAME=capi/%n



When I do a capiinit it says



ERROR: cannot load module kernelcapi



Does there need to be another entry for kernelcapi?





Any help would be greatly appreciated.








___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] CAPI problem under gentoo with AVM C2 - asterisk claims CAPI not installed

2005-11-11 Thread Johan Helsingius
Hi!

I am running Asterisk (1.0.8) on gentoo (2.6.13-gentoo-r5), with AVM C2
ISDN card. capiiinit works OK, capiinfo shows card is up and running
with CAPI OK, but asterisk refuses to load the capi-cm module
(chan_capi-cm, 0.5.4) giving the warning
CAPI not installed, CAPI disabled!.

Any hints of where to look next?

Julf


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI problem under gentoo with AVM C2 - asterisk claims CAPI not installed

2005-11-11 Thread Armin Schindler
 Hi!
 
 I am running Asterisk (1.0.8) on gentoo (2.6.13-gentoo-r5), with AVM C2
 ISDN card. capiiinit works OK, capiinfo shows card is up and running
 with CAPI OK, but asterisk refuses to load the capi-cm module
 (chan_capi-cm, 0.5.4) giving the warning
 CAPI not installed, CAPI disabled!.
 
 Any hints of where to look next?

Any further messages when starting Asterisk with higher verbose level?

Correct permissions to access /dev/capi20 for Asterisk?

Armin

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI problem under gentoo with AVM C2 - asterisk claims CAPI not installed

2005-11-11 Thread Johan Helsingius
Armin Schindler wrote:

 Correct permissions to access /dev/capi20 for Asterisk?

Duh! Of course it had to be something as trivial as that! Thanks!!

Julf
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI - displaying individual MSN

2005-10-18 Thread Stefan Günther
Hi,

I'm currently using chan_capi-cm-0.6, with the following capi.conf:

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

[ISDN1]
msn=8304490
incomingmsn=8304490
isdnmode=msn
group=1
controller=1
softdtmf=1
context=demo
echosquelch=1
echocancel=yes
echotail=64
callgroup=1
devices=2

Each user has a different numer, e.g. 83044910, 83044911, 83044912 and
so on.
This number should appear on the display of the called party, but how do
I configure this?
With the above configuration the display always shows 8304490.
I've tried to change the number in the dialplan, but this doesn't change
anything:

exten = _90[23456789].,1,SetCIDNum(83044912)
exten = _90[23456789].,2,Dial(Capi/g1/${EXTEN:1},30,tr)

If I remove the mns line in the capi.conf or set msn=* or msn=830449*
Asterisk isn't able to open the CAPI channel.

Does anyone have a hint for me?
If yes - THANK YOU ;-)

Stefan

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI - displaying individual MSN

2005-10-18 Thread Peer Oliver Schmidt

Stefan Günther schrieb:


With the above configuration the display always shows 8304490.
I've tried to change the number in the dialplan, but this doesn't change
anything:

exten = _90[23456789].,1,SetCIDNum(83044912)


Try to use SetCallerID instead of SetCIDNum and see if it helps.


exten = _90[23456789].,2,Dial(Capi/g1/${EXTEN:1},30,tr)




--
Best regards

Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI - displaying individual MSN

2005-10-18 Thread Philipp von Klitzing
Hi!

 msn=8304490
 incomingmsn=8304490
 
 Each user has a different numer, e.g. 83044910, 83044911, 83044912 and
 so on.
 This number should appear on the display of the called party, but how do
 I configure this?
 With the above configuration the display always shows 8304490.
 I've tried to change the number in the dialplan, but this doesn't change
 anything:
 
 exten = _90[23456789].,1,SetCIDNum(83044912)
 exten = _90[23456789].,2,Dial(Capi/g1/${EXTEN:1},30,tr)
 
 If I remove the mns line in the capi.conf or set msn=* or msn=830449*
 Asterisk isn't able to open the CAPI channel.

You need to modify incomingmsn= and not msn= for this to work as
expected. Also be aware that often these two settings require different
values for the same meaning, e.g. you might have to add the area prefix
for the msn= setting (40 for Hamburg, 89 for München etc). If however
your Asterisk is behind a PBX then your incoming MSN might only have to be 910, 
911 and 912.

The above applies also to your SetCIDNum statement, it must match a
valid (!) MSN.

Cheers, Philipp


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI - displaying individual MSN

2005-10-18 Thread Armin Schindler
On Tue, 18 Oct 2005, Stefan Günther wrote:
..
 Each user has a different numer, e.g. 83044910, 83044911, 83044912 and
 so on.
 This number should appear on the display of the called party, but how do
 I configure this?
 With the above configuration the display always shows 8304490.
 I've tried to change the number in the dialplan, but this doesn't change
 anything:
 
 exten = _90[23456789].,1,SetCIDNum(83044912)
 exten = _90[23456789].,2,Dial(Capi/g1/${EXTEN:1},30,tr)
 
 If I remove the mns line in the capi.conf or set msn=* or msn=830449*
 Asterisk isn't able to open the CAPI channel.

msn= does not exist anymore, it has no effect.
Use incomingmsn=* to specify which MSN shall be handled by Astreisk.

Are you sure you have PtMP (MSN) connection? When you have numbers like
83044910, 83044911, 83044912,... and the display shows 8304490, then it 
looks like a PtP connection with base number 830449-X.
If thats the case, you should
- switch to isdnmode=did
- SetCIDNum(12), instead of SetCIDNum(83044912)

Armin
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] CAPI problem - need help

2005-08-18 Thread Arik Funke

Armin Schindler wrote:
What kernel do you use? If it's 2.6.10 or newer, then make sure you use new 
chan_capi-cm from sourceforge.net. Older chan_capi is buggy.


Thanks for the hint. I am using kernel 2.6.12. Changing to the 
chan_capi-cm solved the problem... Looks like I could have saved myself 
a lot of work if I had asked earlier. ;-)


Thanks again!

Arik
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI problem - need help

2005-08-17 Thread Arik Funke
I have installed a Fritz card which I use with chan_capi. If the card is 
CALLED, everything works perfectly well.


BUT: If the card is CALLING, it only sends audio but does not receive 
it. I have already changed the card, the remote devices etc. I am 
running out of ideas.


Does anybody know this phenomena? I would really appreciate any ideas I 
could try...


Cheers,
Arik
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI problem - need help

2005-08-17 Thread Armin Schindler
On Wed, 17 Aug 2005, Arik Funke wrote:
 I have installed a Fritz card which I use with chan_capi. If the card is
 CALLED, everything works perfectly well.
 
 BUT: If the card is CALLING, it only sends audio but does not receive it. I
 have already changed the card, the remote devices etc. I am running out of
 ideas.
 
 Does anybody know this phenomena? I would really appreciate any ideas I could
 try...

What kernel do you use? If it's 2.6.10 or newer, then make sure you use new 
chan_capi-cm from sourceforge.net. Older chan_capi is buggy.

Armin

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI Eicon Server bri, extreme noise or gain

2005-07-28 Thread Armin Schindler
On Mon, 25 Jul 2005 [EMAIL PROTECTED] wrote:
 Hello All,
 I installed an eicon diva server.  I have the channels up, however my
 issue is that the sound is unbearable.  The signalling seems to work ok,
 but its as if the gain is so high that its confusing the card.  For
 example, you can hear the ivr prompts, but combined with insane noise
 levels.
 
 Has anyone experienced an issue such as this in the past?

Which version of chan_capi/kernel/divas-driver do you use ?

Armin
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI Eicon Server bri, extreme noise or gain

2005-07-25 Thread gw
Hello All,
I installed an eicon diva server.  I have the channels up, however my
issue is that the sound is unbearable.  The signalling seems to work ok,
but its as if the gain is so high that its confusing the card.  For
example, you can hear the ivr prompts, but combined with insane noise
levels.

Has anyone experienced an issue such as this in the past?

Thanks,
Greg
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] capi or mISDN for passive Fritz!Card PCi

2005-07-22 Thread Eric Bishop
Hi all,

chan someone who has tried BOTH chan_capi and chan_mISDN with a
passive Frtiz!Card PCI comment on one versus the other. Which had
better sound quality.

Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI in PTP mode not answering, dial out fine

2005-07-14 Thread asterisk

Hi list,

I am using Asterisk in a small systems with an AVM C4 card, we first had 
one ISDN line, (ptmp), which we upgraded to

2 ISDN with 1 number (so no DID's) This runs in ptp mode.
Calling out works fine on all 4 channels, but when I call in, I get
*CLI Jul 13 09:44:59 ERROR[13635]: chan_capi.c:1695 pipe_msg: did not 
find device for msn = 299450707


(my number is 0299-450707

The call gets to the C4 card, my kernel logs:

isdn_net: call from 62411 - 0 299450707 ignored
isdn_tty: call from 62411 - 299450707 ignored
capidrv-1: incoming call 62411,1,0,299450707 ignored

my capi.conf:
[interfaces]
isdnmode=ptp
mode=immediate
msn=299450707
incomingmsn=299450707
controller=1,2
softdtmf=1
context=outbound
echosquelch=1
echocancel=yes
echotail=64
callgroup=1
devices=4

extensions.conf (part)
[outbound]
ignorepat = 0

exten = _0.,1,Ringing
exten = _0.,2,Dial(CAPI/299450707:${EXTEN:1})
exten = _0.,3,Congestion

[default]

exten = s,1,Dial(sip/20,25)
exten = s,2,Dial(sip/21,25)

exten = _299450707,1,Goto(s,1)
exten = 0299450707,1,Goto(s,1)
exten =_450707,1,Goto(s,1)
exten = 299450707,1,Goto(s,1)
include = outbound

As you can see I've tried every possible option to get asterisk to match 
the MSN,
but the because the error says no _DEVICE_  found, I don;t think it will 
even make it to the extensions.conf.


I use asterisk 1.07 with chan_capi 0.35

Kind regards,

Joop Marijne

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI in PTP mode not answering, dial out fine

2005-07-14 Thread Armin Schindler
On Thu, 14 Jul 2005, asterisk wrote:
 Hi list,
 
 I am using Asterisk in a small systems with an AVM C4 card, we first had one
 ISDN line, (ptmp), which we upgraded to
 2 ISDN with 1 number (so no DID's) This runs in ptp mode.
 Calling out works fine on all 4 channels, but when I call in, I get
 *CLI Jul 13 09:44:59 ERROR[13635]: chan_capi.c:1695 pipe_msg: did not find
 device for msn = 299450707
...
 my capi.conf:
 [interfaces]
 isdnmode=ptp
 mode=immediate
 msn=299450707
 incomingmsn=299450707
 controller=1,2
 softdtmf=1
 context=outbound

Your context is 'outbound', but

 echosquelch=1
 echocancel=yes
 echotail=64
 callgroup=1
 devices=4
 
 extensions.conf (part)
 [outbound]
 ignorepat = 0
 
 exten = _0.,1,Ringing
 exten = _0.,2,Dial(CAPI/299450707:${EXTEN:1})
 exten = _0.,3,Congestion

here in 'outbound' there is no match to your msn.
 
 As you can see I've tried every possible option to get asterisk to match the
 MSN,
 but the because the error says no _DEVICE_  found, I don;t think it will even
 make it to the extensions.conf.

This message is confusing here, but it seems that the match is not found in
extensions.conf.
 
 I use asterisk 1.07 with chan_capi 0.35

Maybe you want to try chan_capi-cm on sourceforge...

Armin
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI and Caller ID name not showing.

2005-06-29 Thread louis g
I have an Asterisk server connected to ISDN2 lines off a PBX (Avaya) using 4 
port Eicon Diva card. All works fine, but i'd like calls from the PBX to 
Asterisk to show the Caller ID name and not just the number. I know this 
information is being presented by looking through the ISDN trace for the 
Eicon Card. Asterisk trace show dialparties.agi: Caller ID name is '605' 
number is '605'. Can anyone point me in the right direction to get this 
sorted?. It's works with X100P cards :)


_
Winks  nudges are here - download MSN Messenger 7.0 today! 
http://messenger.msn.co.uk


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI and Caller ID name not showing.

2005-06-29 Thread Armin Schindler
On Wed, 29 Jun 2005, louis g wrote:
 I have an Asterisk server connected to ISDN2 lines off a PBX (Avaya) using 4
 port Eicon Diva card. All works fine, but i'd like calls from the PBX to
 Asterisk to show the Caller ID name and not just the number. I know this
 information is being presented by looking through the ISDN trace for the Eicon
 Card. Asterisk trace show dialparties.agi: Caller ID name is '605' number is
 '605'. Can anyone point me in the right direction to get this sorted?. It's
 works with X100P cards :)

What 'name' do you mean? Is it a subaddress?
Please paste an example for that Eicon card trace where you see that name.

Armin

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI and Caller ID name not showing.

2005-06-29 Thread Stefan Gofferje
On 15:54:12 June 29, 2005 Armin Schindler [EMAIL PROTECTED] wrote:
 On Wed, 29 Jun 2005, louis g wrote:
   I have an Asterisk server connected to ISDN2 lines off a PBX
   (Avaya) using 4 port Eicon Diva card. All works fine, but i'd like
   calls from the PBX to Asterisk to show the Caller ID name and not
   just the number. I know this information is being presented by
   looking through the ISDN trace for the Eicon Card. Asterisk trace
   show dialparties.agi: Caller ID name is '605' number is '605'. Can
   anyone point me in the right direction to get this sorted?. It's
 works with X100P cards :)

 What 'name' do you mean? Is it a subaddress?
 Please paste an example for that Eicon card trace where you see that
 name.

His PBX probably transmits the name per UUS1. zaphfc supports this also. I
have a zaphfc card as internal ISDN and connected a Siemens ISDN DECT phone
to it. Now, on incoming calls, the Siemens shows the CallerIDName as set by
Asterisk in the display. zaphfc also supports SendText...

Regards,
Stefan

-- 
  (o_   Stefan Gofferje  | Linux Systems Specialist
  //\   Reg'd Linux User #247167 | Network Security Specialist
  V_/_  Heckler  Koch - the original point and click interface

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI and Caller ID name not showing.

2005-06-29 Thread Christian Händel

Hi,
if you are using the QSIG protocol  for the interconnection between 
Asterisk and the PBX, I have maybe a solution.
for the X100P you are using Zapata driver of asterisk. (with the 
switchtype QSIG right?)

But for the eicon you use the capi module?

Caller Name within QSIG is standardized as Calling Name Identification 
Presentation (CNIP).

 CNIP is implemented in libpri/Zapata but not in the capi of asterisk.
that's because CNIP is not standardized in capi.

But we are lucky: Eicon has made some hacks in his capi driver, so it's 
possible to use CNIP with Eicon-Capi.


I am writing at the moment on the implementation of Eicon-capi-CNIP for 
asterisk. hopefully it will work...


Chris

   zaArmin Schindler wrote:

On Wed, 29 Jun 2005, louis g wrote:


I have an Asterisk server connected to ISDN2 lines off a PBX (Avaya) using 4
port Eicon Diva card. All works fine, but i'd like calls from the PBX to
Asterisk to show the Caller ID name and not just the number. I know this
information is being presented by looking through the ISDN trace for the Eicon
Card. Asterisk trace show dialparties.agi: Caller ID name is '605' number is
'605'. Can anyone point me in the right direction to get this sorted?. It's
works with X100P cards :)



What 'name' do you mean? Is it a subaddress?
Please paste an example for that Eicon card trace where you see that name.

Armin

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI and Caller ID name not showing.

2005-06-29 Thread Armin Schindler
On Wed, 29 Jun 2005, Christian Händel wrote:
 Hi,
 if you are using the QSIG protocol  for the interconnection between Asterisk
 and the PBX, I have maybe a solution.
 for the X100P you are using Zapata driver of asterisk. (with the switchtype
 QSIG right?)
 But for the eicon you use the capi module?
 
 Caller Name within QSIG is standardized as Calling Name Identification
 Presentation (CNIP).
  CNIP is implemented in libpri/Zapata but not in the capi of asterisk.
 that's because CNIP is not standardized in capi.
 
 But we are lucky: Eicon has made some hacks in his capi driver, so it's
 possible to use CNIP with Eicon-Capi.
 
 I am writing at the moment on the implementation of Eicon-capi-CNIP for
 asterisk. hopefully it will work...

That would be great :-)

Armin___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] capi dial in/out configuration

2005-05-27 Thread Ohad.Levy








Hi
all,



I've
recentrly starting to play around with *, when all I wanted is to configure an
fritz ISDN card with [EMAIL PROTECTED]

Currently
I'm stuck at the phase of what do I do with capi after everything is installed.

I'm
trying to understand how to setup incoming and outgoing calls at [EMAIL PROTECTED] since I'm
getting a bit lost with the default dial plan.

It
seems that * answers but disconnect it directly, and I'm unable to setup
outgoing calls.

I
know this is a very general question, but if anyone could give me some pointers
about how to setup capi dial plan, and explain some terms like msn in the
capi.conf file.



My
capi.conf



[EMAIL PROTECTED] asterisk]# cat
capi.conf |grep -v ';'

[general]

nationalprefix=0

internationalprefix=00

rxgain=0.8

txgain=0.8



[interfaces]



msn=50

incomingmsn=*

controller=1

softdtmf=1

accountcode=

context=demo

devices=2



I've
added these two lines the extensions_custom:



s,1,Dial,CAPI/@50:b${EXTEN}|30
always early B3

s,1,Dial,CAPI/@50:${EXTEN}|30|r
no early B3, fake ring indication





when
dialing out I get:



--
Executing Macro(SIP/200-3b6b, dialout-trunk|1|999) in
new stack

 -- Executing
GotoIf(SIP/200-3b6b, fooOhad?4) in new stack

 -- Executing
SetCallerID(SIP/200-3b6b, Ohad Levy) in new stack

 -- Executing
Goto(SIP/200-3b6b, 6) in new stack

 -- Goto (macro-dialout-trunk,s,6)

 -- Executing
SetGroup(SIP/200-3b6b, OUT_1) in new stack

 -- Executing
CheckGroup(SIP/200-3b6b, ) in new stack

 -- Executing
SetVar(SIP/200-3b6b, DIAL_NUMBER=999) in new stack

 -- Executing
SetVar(SIP/200-3b6b, DIAL_TRUNK=1) in new stack

 -- Executing
AGI(SIP/200-3b6b, fixlocalprefix) in new stack

 -- Launched
AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix

 -- AGI Script
fixlocalprefix completed, returning 0

 -- Executing
Dial(SIP/200-3b6b, /999) in new stack

 == Everyone is busy/congested
at this time

 -- Executing
NoOp(SIP/200-3b6b, dial failed) in new stack

 -- Executing
Macro(SIP/200-3b6b, outisbusy) in new stack

 -- Executing
Playback(SIP/200-3b6b, allison7/all-circuits-busy-now) in
new stack

 -- Playing 'allison7/all-circuits-busy-now'
(language 'en')

 == Spawn extension (macro-outisbusy,
s, 1) exited non-zero on 'SIP/200-3b6b' in macro 'outisbusy'

 == Spawn extension (from-internal,
, 2) exited non-zero on 'SIP/200-3b6b'

 -- Executing
Macro(SIP/200-3b6b, hangupcall) in new stack

 -- Executing
ResetCDR(SIP/200-3b6b, w) in new stack == Starting CAPI[contr1/8856224]/0
at demo,8856224,1 failed so falling back to exten 's'

 == Starting CAPI[contr1/8856224]/0
at demo,s,1 still failed so falling back to context 'default'

 -- Executing
Playback(CAPI[contr1/8856224]/0, vm-goodbye) in new
stack

 -- started
pbx on channel (callgroup=0)!

 -- Playing 'vm-goodbye'
(language 'en')

 -- Executing
Macro(CAPI[contr1/8856224]/0, hangupcall) in new stack

 -- Executing
ResetCDR(CAPI[contr1/8856224]/0, w) in new stack

 -- Executing
NoCDR(CAPI[contr1/8856224]/0, ) in new stack

 -- Executing
Wait(CAPI[contr1/8856224]/0, 5) in new stack

 -- Executing
Hangup(CAPI[contr1/8856224]/0, ) in new stack

 == Spawn extension (macro-hangupcall,
s, 4) exited non-zero on 'CAPI[contr1/8856224]/0' in macro 'hangupcall'

 == Spawn extension (default,
s, 2) exited non-zero on 'CAPI[contr1/8856224]/0'

 -- Executing
NoCDR(SIP/200-3b6b, ) in new stack

 -- Executing
Wait(SIP/200-3b6b, 5) in new stack

 == Spawn extension (macro-hangupcall,
s, 3) exited non-zero on 'SIP/200-3b6b' in macro 'hangupcall'

 == Spawn extension (from-internal,
h, 1) exited non-zero on 'SIP/200-3b6b'





When
receiving a call:

==
Starting CAPI[contr1/myisdn#]/0 at demo, myisdn#,1 failed so
falling back to exten 's'

 == Starting CAPI[contr1/myisdn#]/0
at demo,s,1 still failed so falling back to context 'default'

 -- Executing
Playback(CAPI[contr1/myisdn#]/0, vm-goodbye) in
new stack

 -- started
pbx on channel (callgroup=0)!

 -- Playing 'vm-goodbye'
(language 'en')

 -- Executing
Macro(CAPI[contr1/myisdn#]/0, hangupcall) in
new stack

 -- Executing
ResetCDR(CAPI[contr1/myisdn#]/0, w) in new
stack

 -- Executing
NoCDR(CAPI[contr1/myisdn#]/0, ) in new stack

 -- Executing
Wait(CAPI[contr1/myisdn#]/0, 5) in new stack

 -- Executing
Hangup(CAPI[contr1/myisdn#]/0, ) in new stack

 == Spawn extension (macro-hangupcall,
s, 4) exited non-zero on 'CAPI[contr1/myisdn#]/0' in macro 'hangupcall'

 == Spawn extension (default,
s, 2) exited non-zero on 'CAPI[contr1/myisdn#]/0'





Thanks
a lot,

Ohad






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

AW: [Asterisk-Users] CAPI on ptp with variable length digits inphonenumber: SOLUTION for EICON

2005-05-09 Thread Sebastian Buntin
 
the interesting fact is, it works. I dunno why. but it works :o


-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von bladerunner
Gesendet: Freitag, 6. Mai 2005 15:56
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [Asterisk-Users] CAPI on ptp with variable length digits 
inphonenumber: SOLUTION for EICON

but with this solution you will not be able to receive calls with less than 
two DID-digits (like call to 123-0 where 123 is head number). it will wait 
for exactly two digits before answering (at least in the last version of the 
firmware and chan_capi i tried).

regards,

Am Freitag, 6. Mai 2005 15:17 schrieb Sebastian Buntin:
 Hello!

 I finally found a working solution.
 calling
 divactrl with the parameter -n [0..20] gives the DID-length
 means, if you wanna have 123-XXX  in digit-wise mode, then call

 divactrl load -c 1 -n 3 -f ETSI

 and the card will wait for n digits.


 regards,
 Sebastian


 -Ursprüngliche Nachricht-
 Von: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Im Auftrag von bladerunner
 Gesendet: Freitag, 6. Mai 2005 14:01
 An: Asterisk Users Mailing List - Non-Commercial Discussion
 Betreff: Re: [Asterisk-Users] CAPI on ptp with variable length digits in
 phonenumber

 hi again,

 just ignore my mentioning of the sirrix-cards, just realised you have a
 PRI, i overread it and thought you had a BRI. so i think your last hope is
 a zaptel-card.

 regards,

 Am Freitag, 6. Mai 2005 13:04 schrieb Sebastian Buntin:
  Hi!
 
  we have a german PtP PRI connection here.
  our old telephone system was programmed to accept digits by variable
  length.
  so our MSN, assigned my telco is, lets say: 123
  and we can use first digit from 0-4. and every further digit like we
  want.
  means:
 
  123-1
  123-2345
  123-44
  till
  123-499
  but not 123-5...
 
  I'm using an Eicon diva server PRI 23M with chan_capi.
 
  my problem is dialing IN.
 
  block mode works perfect. (when the whole number is sent as a block)
 
  I just add
 
  exten = 123114,1,Dial(SIP/blahblah)
 
  works.
 
  but if someone dials digit-wise, lets say 123114,
  asterisk starts scanning the dialplan after 1231.
  doesnt find an extension and exits.
  even using
  exten = s,1,DigitTimeout,4
  as first line in the dialin-extension won't help.
 
  so, I need to find a way that asterisk collects the digits until it has
  a matching one.
  p.ex. wait scanning the extensions till the caller typed 123114.
  I can live with fixed length extensions. means, always wait for 3
  digits.
 
 
  thanks for help..
 
  Sebastian
  ___
  Asterisk-Users mailing list
  Asterisk-Users@lists.digium.com
  http://lists.digium.com/mailman/listinfo/asterisk-users
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users

 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI on ptp with variable length digits in phone number

2005-05-06 Thread Sebastian Buntin


Hi!

we have a german PtP PRI connection here.
our old telephone system was programmed to accept digits by variable
length.
so our MSN, assigned my telco is, lets say: 123
and we can use first digit from 0-4. and every further digit like we
want.
means:

123-1
123-2345
123-44
till
123-499
but not 123-5...

I'm using an Eicon diva server PRI 23M with chan_capi.

my problem is dialing IN.

block mode works perfect. (when the whole number is sent as a block)

I just add

exten = 123114,1,Dial(SIP/blahblah)

works.

but if someone dials digit-wise, lets say 123114,
asterisk starts scanning the dialplan after 1231.
doesnt find an extension and exits.
even using 
exten = s,1,DigitTimeout,4
as first line in the dialin-extension won't help.

so, I need to find a way that asterisk collects the digits until it has
a matching one.
p.ex. wait scanning the extensions till the caller typed 123114.
I can live with fixed length extensions. means, always wait for 3
digits.


thanks for help..

Sebastian
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI on ptp with variable length digits in phone number

2005-05-06 Thread bladerunner
hi back,

we had this sort of problems (and some other ones... sigh) with an eicon diva 
 chan_capi in austria.

rant mode=on

unfortunately we never got around to fix this spaghetti of a code in 
chan_capi.c to work as intended 

(it takes the first session_setup it gets from the line and ignores, that per 
specification of isdn you can send digits for DID _after_ the setup of the 
connection too. there should be a timer with timeout for waiting for those 
digits. we tried to implement a separate thread into chan_capi to handle this 
timeout, but the code and the variable-naming is so obscure, we never got 
arround to finding let alone fixing a invalid pointer not freed bug in our 
hack of this hack).

rant mode=off

as a last desperate try we got a sirrix-card (search on the wiki for sirrix) 
and i had some very good calls with sirrix' development department, who fixed 
all problems in the isdn-layer and the asterisk-channel (chan_sirrix) for us. 
works perfectly.

sorry i have no immediate solution, but i dumped the avm and eicon cards 
completely in favor of the sirrix ones.

Am Freitag, 6. Mai 2005 13:04 schrieb Sebastian Buntin:
 Hi!

 we have a german PtP PRI connection here.
 our old telephone system was programmed to accept digits by variable
 length.
 so our MSN, assigned my telco is, lets say: 123
 and we can use first digit from 0-4. and every further digit like we
 want.
 means:

 123-1
 123-2345
 123-44
 till
 123-499
 but not 123-5...

 I'm using an Eicon diva server PRI 23M with chan_capi.

 my problem is dialing IN.

 block mode works perfect. (when the whole number is sent as a block)

 I just add

 exten = 123114,1,Dial(SIP/blahblah)

 works.

 but if someone dials digit-wise, lets say 123114,
 asterisk starts scanning the dialplan after 1231.
 doesnt find an extension and exits.
 even using
 exten = s,1,DigitTimeout,4
 as first line in the dialin-extension won't help.

 so, I need to find a way that asterisk collects the digits until it has
 a matching one.
 p.ex. wait scanning the extensions till the caller typed 123114.
 I can live with fixed length extensions. means, always wait for 3
 digits.


 thanks for help..

 Sebastian
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


pgpB1mKAGhmIe.pgp
Description: PGP signature
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] CAPI on ptp with variable length digits in phone number

2005-05-06 Thread bladerunner
hi again,

just ignore my mentioning of the sirrix-cards, just realised you have a PRI, i 
overread it and thought you had a BRI. so i think your last hope is a 
zaptel-card.

regards,

Am Freitag, 6. Mai 2005 13:04 schrieb Sebastian Buntin:
 Hi!

 we have a german PtP PRI connection here.
 our old telephone system was programmed to accept digits by variable
 length.
 so our MSN, assigned my telco is, lets say: 123
 and we can use first digit from 0-4. and every further digit like we
 want.
 means:

 123-1
 123-2345
 123-44
 till
 123-499
 but not 123-5...

 I'm using an Eicon diva server PRI 23M with chan_capi.

 my problem is dialing IN.

 block mode works perfect. (when the whole number is sent as a block)

 I just add

 exten = 123114,1,Dial(SIP/blahblah)

 works.

 but if someone dials digit-wise, lets say 123114,
 asterisk starts scanning the dialplan after 1231.
 doesnt find an extension and exits.
 even using
 exten = s,1,DigitTimeout,4
 as first line in the dialin-extension won't help.

 so, I need to find a way that asterisk collects the digits until it has
 a matching one.
 p.ex. wait scanning the extensions till the caller typed 123114.
 I can live with fixed length extensions. means, always wait for 3
 digits.


 thanks for help..

 Sebastian
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


pgpNP7ftIvREg.pgp
Description: PGP signature
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] CAPI on ptp with variable length digits in phonenumber: SOLUTION for EICON

2005-05-06 Thread Sebastian Buntin
 

Hello!

I finally found a working solution.
calling 
divactrl with the parameter -n [0..20] gives the DID-length
means, if you wanna have 123-XXX  in digit-wise mode, then call 

divactrl load -c 1 -n 3 -f ETSI

and the card will wait for n digits.


regards,
Sebastian


-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von bladerunner
Gesendet: Freitag, 6. Mai 2005 14:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [Asterisk-Users] CAPI on ptp with variable length digits in 
phonenumber

hi again,

just ignore my mentioning of the sirrix-cards, just realised you have a PRI, i 
overread it and thought you had a BRI. so i think your last hope is a 
zaptel-card.

regards,

Am Freitag, 6. Mai 2005 13:04 schrieb Sebastian Buntin:
 Hi!

 we have a german PtP PRI connection here.
 our old telephone system was programmed to accept digits by variable
 length.
 so our MSN, assigned my telco is, lets say: 123
 and we can use first digit from 0-4. and every further digit like we
 want.
 means:

 123-1
 123-2345
 123-44
 till
 123-499
 but not 123-5...

 I'm using an Eicon diva server PRI 23M with chan_capi.

 my problem is dialing IN.

 block mode works perfect. (when the whole number is sent as a block)

 I just add

 exten = 123114,1,Dial(SIP/blahblah)

 works.

 but if someone dials digit-wise, lets say 123114,
 asterisk starts scanning the dialplan after 1231.
 doesnt find an extension and exits.
 even using
 exten = s,1,DigitTimeout,4
 as first line in the dialin-extension won't help.

 so, I need to find a way that asterisk collects the digits until it has
 a matching one.
 p.ex. wait scanning the extensions till the caller typed 123114.
 I can live with fixed length extensions. means, always wait for 3
 digits.


 thanks for help..

 Sebastian
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI on ptp with variable length digits in phonenumber: SOLUTION for EICON

2005-05-06 Thread bladerunner
but with this solution you will not be able to receive calls with less than 
two DID-digits (like call to 123-0 where 123 is head number). it will wait 
for exactly two digits before answering (at least in the last version of the 
firmware and chan_capi i tried).

regards,

Am Freitag, 6. Mai 2005 15:17 schrieb Sebastian Buntin:
 Hello!

 I finally found a working solution.
 calling
 divactrl with the parameter -n [0..20] gives the DID-length
 means, if you wanna have 123-XXX  in digit-wise mode, then call

 divactrl load -c 1 -n 3 -f ETSI

 and the card will wait for n digits.


 regards,
 Sebastian


 -Ursprüngliche Nachricht-
 Von: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Im Auftrag von bladerunner
 Gesendet: Freitag, 6. Mai 2005 14:01
 An: Asterisk Users Mailing List - Non-Commercial Discussion
 Betreff: Re: [Asterisk-Users] CAPI on ptp with variable length digits in
 phonenumber

 hi again,

 just ignore my mentioning of the sirrix-cards, just realised you have a
 PRI, i overread it and thought you had a BRI. so i think your last hope is
 a zaptel-card.

 regards,

 Am Freitag, 6. Mai 2005 13:04 schrieb Sebastian Buntin:
  Hi!
 
  we have a german PtP PRI connection here.
  our old telephone system was programmed to accept digits by variable
  length.
  so our MSN, assigned my telco is, lets say: 123
  and we can use first digit from 0-4. and every further digit like we
  want.
  means:
 
  123-1
  123-2345
  123-44
  till
  123-499
  but not 123-5...
 
  I'm using an Eicon diva server PRI 23M with chan_capi.
 
  my problem is dialing IN.
 
  block mode works perfect. (when the whole number is sent as a block)
 
  I just add
 
  exten = 123114,1,Dial(SIP/blahblah)
 
  works.
 
  but if someone dials digit-wise, lets say 123114,
  asterisk starts scanning the dialplan after 1231.
  doesnt find an extension and exits.
  even using
  exten = s,1,DigitTimeout,4
  as first line in the dialin-extension won't help.
 
  so, I need to find a way that asterisk collects the digits until it has
  a matching one.
  p.ex. wait scanning the extensions till the caller typed 123114.
  I can live with fixed length extensions. means, always wait for 3
  digits.
 
 
  thanks for help..
 
  Sebastian
  ___
  Asterisk-Users mailing list
  Asterisk-Users@lists.digium.com
  http://lists.digium.com/mailman/listinfo/asterisk-users
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users

 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


pgppedy6biZ82.pgp
Description: PGP signature
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] capi problem with dialout

2005-04-21 Thread Pawe Staszewski

  
  

  Hello 



  I have capi 0.3.5 with fritz card and:



  When i try to dialout i have this message in from capi debug :



  CAPI Debugging Enabled


  CLI -- Executing AnswerSIP/478-3f3f  in new stack


  -- Executing DialSIP/478-3f3f CAPI/@7523071:b7522333 in new stack


  -- data = ""


  -- capi request omsn = @7523071


  == found capi with omsn = 7523071


  == CAPI Call CAPIcontr1/7523071/0 with B3 -- Called @7523071:b7522333


  -- CONNECT_CONF ID=001 0x0004 LEN=0014


  Controller/PLCI/NCCI = 0x101


  Info = 0x0



  == received CONNECT_CONF PLCI = 0x101 INFO = 0


  -- INFO_IND ID=001 0x0b97 LEN=0017


  Controller/PLCI/NCCI = 0x101


  InfoNumber = 0x8


  InfoElement = 81 81



  -- DISCONNECT_IND ID=001 0x0b98 LEN=0014


  Controller/PLCI/NCCI = 0x101


  Reason = 0x3481



  == DISCONNECT_IND PLCI=0x101 REASON=0x3481


  == No one is available to answer at this time




  I have one msn number. 7523071


  extensions.conf



  exten = _0.1Answer


  exten = _0.2DialCAPI/@7523071:bEXTEN:1



  capi.conf


  general


  mode=immediate


  nationalprefix=0


  internationalprefix=00


  rxgain=0.8


  txgain=0.8



  interfaces



  isdnmode=multipoint


  msn=7523071


  incomingmsn=


  controller=1


  softdtmf=1


  context=from-isdn


  echosquelch=1


  echocancel=1


  echotail=64


  callgroup=1


  deflect=478


  devices=2




  Please help me i search all the google and i have nothing :


  Best RegardsPawe StaszewskiART-COM+48327522333+480609183038

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] capi problem with dialout

2005-04-21 Thread Michael Bielicki
Since capi is not even really supported by the guy who wrote it I'd
suggest you get a zaphfc card, like the I-tec ISDN-128, which could
simplify your task a lot.
:)

cheers Micha


On 4/21/05, Pawe Staszewski [EMAIL PROTECTED] wrote:
  
  Hello 
   
  I have capi 0.3.5 with fritz card and: 
   
  When i try to dialout i have this message in from capi debug : 
   
  CAPI Debugging Enabled 
  *CLI -- Executing Answer(SIP/478-3f3f, ) in new stack 
  -- Executing Dial(SIP/478-3f3f, CAPI/@7523071:b7522333) in new
 stack 
  -- data = @7523071:b7522333 
  -- capi request omsn = @7523071 
== found capi with omsn = 7523071 
== CAPI Call CAPI[contr1/7523071]/0 with B3-- Called
 @7523071:b7522333 
  -- CONNECT_CONF ID=001 #0x0004 LEN=0014 
Controller/PLCI/NCCI= 0x101 
Info= 0x0 
   
== received CONNECT_CONF PLCI = 0x101 INFO = 0 
  -- INFO_IND ID=001 #0x0b97 LEN=0017 
Controller/PLCI/NCCI= 0x101 
InfoNumber  = 0x8 
InfoElement = 81 81 
   
  -- DISCONNECT_IND ID=001 #0x0b98 LEN=0014 
Controller/PLCI/NCCI= 0x101 
Reason  = 0x3481 
   
== DISCONNECT_IND PLCI=0x101 REASON=0x3481 
== No one is available to answer at this time 
   
   
  I have one msn number. 7523071 
  extensions.conf 
   
  exten = _0.,1,Answer 
  exten = _0.,2,Dial,CAPI/@7523071:b${EXTEN:1} 
   
  capi.conf 
  [general] 
  mode=immediate 
  nationalprefix=0 
  internationalprefix=00 
  rxgain=0.8 
  txgain=0.8 
   
  [interfaces] 
   
  isdnmode=multipoint 
  msn=7523071 
  incomingmsn=* 
  controller=1 
  softdtmf=1 
  context=from-isdn 
  ;echosquelch=1 
  echocancel=1 
  ;echotail=64 
  ;callgroup=1 
  ;deflect=478 
  devices=2 
   
   
  Please help me i search all the google and i have nothing :) 
   
 Best Regards
 Pawe Staszewski
 ART-COM
 +48327522333
 +480609183038
 
 
  
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
   
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 


-- 
Michal Bielicki
http://www.aefirion.org/
http://www.asterisk.com.pl/
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] capi problem with dialout

2005-04-21 Thread Dave Cotton
On Thu, 2005-04-21 at 12:09 +0200, Pawe Staszewski wrote:
 Hello 
  
 I have capi 0.3.5 with fritz card and: 
  
 When i try to dialout i have this message in from capi debug : 
  
 CAPI Debugging Enabled 
 *CLI -- Executing Answer(SIP/478-3f3f, ) in new stack 
 -- Executing Dial(SIP/478-3f3f, CAPI/@7523071:b7522333) in new
 stack 
 -- data = @7523071:b7522333 
 -- capi request omsn = @7523071 
   == found capi with omsn = 7523071 
   == CAPI Call CAPI[contr1/7523071]/0 with B3-- Called
 @7523071:b7522333 
 -- CONNECT_CONF ID=001 #0x0004 LEN=0014 
   Controller/PLCI/NCCI= 0x101 
   Info= 0x0 
  
   == received CONNECT_CONF PLCI = 0x101 INFO = 0 
 -- INFO_IND ID=001 #0x0b97 LEN=0017 
   Controller/PLCI/NCCI= 0x101 
   InfoNumber  = 0x8 
   InfoElement = 81 81 
  
 -- DISCONNECT_IND ID=001 #0x0b98 LEN=0014 
   Controller/PLCI/NCCI= 0x101 
   Reason  = 0x3481 
  
   == DISCONNECT_IND PLCI=0x101 REASON=0x3481 
   == No one is available to answer at this time 
  

How changing from CAPI to a zaphfc card will correct this error I don't
know, and problably neither does the person who suggested it.

REASON 0x3481 is Unallocated (unassigned) number. = Wrong number.

-- 
Dave Cotton [EMAIL PROTECTED]

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] capi problem with dialout

2005-04-21 Thread Robert Webb
SNIP
  == DISCONNECT_IND PLCI=0x101 REASON=0x3481 
  == No one is available to answer at this time 
 
How changing from CAPI to a zaphfc card will correct 
this error I don't
know, and problably neither does the person who 
suggested it.

REASON 0x3481 is Unallocated (unassigned) number. = 
Wrong number.

--
Dave Cotton [EMAIL PROTECTED]

Just as a shot in the dark, but does the telco maybe 
require  10 digit dialing for ISDN??
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Odp: Re: [Asterisk-Users] capi problem with dialout

2005-04-21 Thread Pawe Staszewski

  
  
  Hello 


I live in poland and : 

local numbers are: 752 7 digits

zone prefix: 32

country prefix: 48


And i must add that i am behind a local PBX Alcatel 4200E

Configured isdn port with msn 7523071


Why dial in is working but dial-out not ...  


And: I can dial-in from outside some debug from capi :

-- CONNECT_IND ID=001 0x0e29 LEN=0045

Controller/PLCI/NCCI = 0x101

CIPValue = 0x10

CalledPartyNumber = 81153

CallingPartyNumber = 09 80172

CalledPartySubaddress = default

CallingPartySubaddress = default

BC = 80 90 a3

LLC = default

HLC = 91 81

AdditionalInfo

BChannelinformation = 00 00

Keypadfacility = default

Useruserdata = 04

Facilitydataarray = default


== CONNECT_IND PLCI=0x101DID=153CID=172CIP=0x10CONTROLLER=0x1

-- started pbx on channel callgroup=0

-- INFO_IND ID=001 0x0e2a LEN=0016

Controller/PLCI/NCCI = 0x101

InfoNumber = 0x7e

InfoElement = 04


-- INFO_IND ID=001 0x0e2b LEN=0019

Controller/PLCI/NCCI = 0x101

InfoNumber = 0x70

InfoElement = 81153


-- INFO_IND ID=001 0x0e2c LEN=0016

Controller/PLCI/NCCI = 0x101

InfoNumber = 0x18

InfoElement = 89


-- ALERT_CONF ID=001 0x0e29 LEN=0014

Controller/PLCI/NCCI = 0x101

Info = 0x0


== Starting CAPIcontr1/153/6 at from-isdn1531 failed so falling back to exten s

-- Executing SetLanguageCAPIcontr1/153/6 en in new stack

-- Executing DialCAPIcontr1/153/6 SIP/478 in new stack

Were at 195.205.186.7 port 10786

Answering with preferred capability 0x4 ulaw

Answering with preferred capability 0x2 gsm

Answering with non-codec capability 0x1 telephone-event

12 headers 11 lines

Reliably Transmitting:

INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0

Via: SIP/2.0/UDP 195.205.186.7:5060branch=z9hG4bK4541e422

From: 172 sip:[EMAIL PROTECTED]tag=as24721ef0

To: sip:[EMAIL PROTECTED]:5060

Contact: sip:[EMAIL PROTECTED]

Call-ID: [EMAIL PROTECTED]

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Date: Thu 21 Apr 2005 14:03:36 GMT

Allow: INVITE ACK CANCEL OPTIONS BYE REFER

Content-Type: application/sdp

Content-Length: 241


v=0

o=root 10839 10839 IN IP4 195.205.186.7

s=session

c=IN IP4 195.205.186.7

t=0 0

m=audio 10786 RTP/AVP 0 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

no NAT to 10.0.230.14:5060

-- Called 478



Sip read:

SIP/2.0 100 Trying

To: sip:[EMAIL PROTECTED]:5060

From: 172sip:[EMAIL PROTECTED]tag=as24721ef0

Via: SIP/2.0/UDP 195.205.186.7:5060branch=z9hG4bK4541e422received=195.205.186.7

Call-ID: [EMAIL PROTECTED]

CSeq: 102 INVITE

Contact: sip:10.0.230.14:5060

User-Agent: Firefly

Content-Length: 0



9 headers 0 lines



Sip read:

SIP/2.0 180 Ringing

To: sip:[EMAIL PROTECTED]:5060tag=c84d4d07

From: 172sip:[EMAIL PROTECTED]tag=as24721ef0

Via: SIP/2.0/UDP 195.205.186.7:5060branch=z9hG4bK4541e422received=195.205.186.7

Call-ID: [EMAIL PROTECTED]

CSeq: 102 INVITE

Contact: sip:10.0.230.14:5060

User-Agent: Firefly

Content-Length: 0



9 headers 0 lines

-- SIP/478-2750 is ringing


-- INFO_IND ID=001 0x0e2d LEN=0017

Controller/PLCI/NCCI = 0x101

InfoNumber = 0x8

InfoElement = 81 90


-- DISCONNECT_IND ID=001 0x0e2e LEN=0014

Controller/PLCI/NCCI = 0x101

Reason = 0x3490


== DISCONNECT_IND PLCI=0x101 REASON=0x3490

Reliably Transmitting:

CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0

Via: SIP/2.0/UDP 195.205.186.7:5060branch=z9hG4bK4541e422

From: 172 sip:[EMAIL PROTECTED]tag=as24721ef0

To: sip:[EMAIL PROTECTED]:5060

Contact: sip:[EMAIL PROTECTED]

Call-ID: [EMAIL PROTECTED]

CSeq: 102 CANCEL

User-Agent: Asterisk PBX

Content-Length: 0


no NAT to 10.0.230.14:5060

Scheduling destruction of call [EMAIL PROTECTED] in 15000 ms

== Spawn extension from-isdn s 2 exited non-zero on CAPIcontr1/153/6



Sip read:

SIP/2.0 200 OK

To: sip:[EMAIL PROTECTED]:5060tag=c84d4d07

From: 172 sip:[EMAIL PROTECTED]tag=as24721ef0

Via: SIP/2.0/UDP 

Re: Odp: Re: [Asterisk-Users] capi problem with dialout

2005-04-21 Thread Peer Oliver Schmidt
Pawe Staszewski wrote:
Hello
I live in poland and :)
local numbers are: 752 (7 digits)
zone prefix: 32
country prefix: 48
And i must add that i am behind a local PBX (Alcatel 4200E)
Configured isdn port with msn 7523071
Why dial in is working but dial-out not ... ??
maybe your local PBX requires a 0 in front for an outside line?
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] capi problem with dialout

2005-04-21 Thread Pawe Staszewski

  
  
  Hello


i try with 0 and

-- Executing AnswerSIP/478-c9a2  in new stack

-- Executing DialSIP/478-c9a2 CAPI/7523071:07522333 in new stack

-- data = ""

-- capi request omsn = 7523071

== found capi with omsn = 7523071

== CAPI Call CAPIcontr1/7523071/7 -- Called 7523071:07522333

-- CONNECT_CONF ID=001 0x0b3f LEN=0014

Controller/PLCI/NCCI = 0x101

Info = 0x0


== received CONNECT_CONF PLCI = 0x101 INFO = 0

-- INFO_IND ID=001 0x1987 LEN=0017

Controller/PLCI/NCCI = 0x101

InfoNumber = 0x8

InfoElement = 81 81


-- DISCONNECT_IND ID=001 0x1988 LEN=0014

Controller/PLCI/NCCI = 0x101

Reason = 0x3481


== DISCONNECT_IND PLCI=0x101 REASON=0x3481



[EMAIL PROTECTED] 04/21/05 5:24 pm PaweStaszewski wrote:HelloI live in poland and :local numbers are: 752 7 digitszone prefix: 32country prefix: 48And i must add that i am behind a local PBX Alcatel 4200EConfigured isdn port with msn 7523071Why dial in is working but dial-out not ... maybe your local PBX requires a 0 in front for an outside line--Best regardsPeer Oliver SchmidtPGP Key ID: 0x83E1C2EAAsterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users

  

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] capi problem with dialout

2005-04-21 Thread Peer Oliver Schmidt
Pawe Staszewski wrote:
Hello
i try with 0 and
   -- Executing Answer(SIP/478-c9a2, ) in new stack
-- Executing Dial(SIP/478-c9a2, CAPI/7523071:07522333) in new stack
-- data = 7523071:07522333
-- capi request omsn = 7523071
  == found capi with omsn = 7523071
  == CAPI Call CAPI[contr1/7523071]/7 -- Called 7523071:07522333
-- CONNECT_CONF ID=001 #0x0b3f LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0
  == received CONNECT_CONF PLCI = 0x101 INFO = 0
-- INFO_IND ID=001 #0x1987 LEN=0017
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8
  InfoElement = 81 81
-- DISCONNECT_IND ID=001 #0x1988 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Reason  = 0x3481
Before going any further with Asterisk, try to verify your CAPI setup is
able to dial out. Install something like capi4hylafax, and see if you
can dialout. If that works come back to asterisk and apply what you have
learned.
chan_capi works at least as reliable as the bristuff. I am running a AVM
C4 plus a HFC-S based card in my asterisk server. The CAPI stuff got me
started, bristuff came later.
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] capi segfault when incoming call is answered

2005-04-20 Thread Jason Williams
On 4/7/05, Thomas Andrews [EMAIL PROTECTED] wrote:
 On Thu, Apr 07, 2005 at 10:15:09AM +0200, Thomas Andrews wrote:
 
  I have a Fritz! card set up to use capi, however when incoming calls to
  the card are answered, asterisk segfaults.
 

Have you tried a make clean then make install in the chan_capi source
directory make sure the header files are built correctly.


Jason
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


RE: [Asterisk-Users] capi segfault when incoming call is answered

2005-04-20 Thread Ivan Meic (Vox Mundi)
  On Thu, Apr 07, 2005 at 10:15:09AM +0200, Thomas Andrews wrote:
 
   I have a Fritz! card set up to use capi, however when incoming calls
to
   the card are answered, asterisk segfaults.
 

 Have you tried a make clean then make install in the chan_capi source
 directory make sure the header files are built correctly.

I'm not totaly sure, but I think I had the same problem
when I upgraded from capi4k-utils-2004-10-06.tar.gz to a newer version.
As soon as I downgraded, it started working normally again.

Good luck,

Ivan

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] capi segfault when incoming call is answered

2005-04-07 Thread Thomas Andrews
I have a Fritz! card set up to use capi, however when incoming calls to
the card are answered, asterisk segfaults. Here is the output of gdb:

#0  0x4014f7af in memcpy () from /lib/tls/libc.so.6
#1  0x081316b0 in ?? ()
#2  0x08130680 in ?? ()
#3  0x40432da6 in pipe_msg (PLCI=257, CMSG=0x405052c0) at chan_capi.c:1560
#4  0x40436f90 in capi_handle_msg (CMSG=0x101) at chan_capi.c:2379
#5  0x404362f7 in do_monitor (data=0x0) at chan_capi.c:2404
#6  0x400229b4 in start_thread () from /lib/tls/libpthread.so.0
#7  0x in ?? ()

The problem is at line 1560 in chan_capi.c:

memcpy(b3buf[AST_FRIENDLY_OFFSET],(char 
*)DATA_B3_IND_DATA(CMSG),DATA_B3_IND_DATALENGTH(CMSG));

I'm using chan_capi-0.3.5 with the patch from
http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2

(A similar thing happens when I make outgoing calls via the Fritz! card
- it segfaults as soon as the phone on the other end starts ringing)

Here's a bit more info from gdb:

8-8---8--

#0  0x4014f7af in memcpy () from /lib/tls/libc.so.6
No symbol table info available.
#1  0x081316b0 in ?? ()
No symbol table info available.
#2  0x08130680 in ?? ()
No symbol table info available.
#3  0x40432da6 in pipe_msg (PLCI=257, CMSG=0x405052c0) at
chan_capi.c:1560
p = (struct capi_pipe *) 0x1e
CMSG2 = {ApplId = 1, Command = 131 '\203', Subcommand = 131
'\203', Messagenumber = 202, adr = {adrController = 131585, 
adrPLCI = 131585, adrNCCI = 131585}, AdditionalInfo = CAPI_COMPOSE,
B1configuration = 0x0, B1protocol = 0, B2configuration = 0x0, 
  B2protocol = 0, B3configuration = 0x0, B3protocol = 0, BC = 0x0,
BChannelinformation = 0x0, BProtocol = CAPI_COMPOSE, 
  CalledPartyNumber = 0x0, CalledPartySubaddress = 0x0,
CallingPartyNumber = 0x0, CallingPartySubaddress = 0x0, CIPmask = 0, 
  CIPmask2 = 0, CIPValue = 0, Class = 0, ConnectedNumber = 0x0,
ConnectedSubaddress = 0x0, Data32 = 0, Data64 = 0, DataHandle = 0, 
  DataLength = 0, FacilityConfirmationParameter = 0x0, Facilitydataarray
= 0x0, FacilityIndicationParameter = 0x0, 
  FacilityRequestParameter = 0x0, FacilityResponseParameters = 0x0,
FacilitySelector = 0, Flags = 0, Function = 0, HLC = 0x0, Info = 0, 
  InfoElement = 0x0, InfoMask = 0, InfoNumber = 0, Keypadfacility = 0x0,
LLC = 0x0, ManuData = 0x0, ManuID = 0, NCPI = 0x0, Reason = 0, 
  Reason_B3 = 0, Reject = 0, Useruserdata = 0x0, SendingComplete = 0x0,
Data = 0xc Address 0xc out of bounds, l = 1, p = 1078220081, 
  par = 0x40449700 \f, m = 0x0, buf = '\0' repeats 179 times}
error = 160
fr = {frametype = 4, subclass = 4, datalen = 0, samples = 128,
mallocd = 192, offset = 0, src = 0x4020aebc Ä}\023, data = 0x41, 
  delivery = {tv_sec = 135278508, tv_usec = 135457433}, prev =
0x818b820, next = 0x1}
b3buf = [EMAIL PROTECTED], '\0' repeats 36 times,
[EMAIL PROTECTED]
@[EMAIL PROTECTED]@[EMAIL PROTECTED]@, '\0' repeats 16 times,
N¢C@, '\0' repeats 60 times, \n¼\037@, '\0' repeats 16 times,
¼® @[EMAIL PROTECTED]@[EMAIL PROTECTED] @`UP@/[EMAIL PROTECTED]@[EMAIL 
PROTECTED]
@[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL 
PROTECTED]@`UP@...
j = 30
b3len = 0
dtmf = 30 '\036'
dtmflen = 1079005888
rxavg = 0
txavg = 0

8-8---8--
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] capi segfault when incoming call is answered

2005-04-07 Thread Thomas Andrews
On Thu, Apr 07, 2005 at 10:15:09AM +0200, Thomas Andrews wrote:

 I have a Fritz! card set up to use capi, however when incoming calls to
 the card are answered, asterisk segfaults.

Just for the record, my capi.conf looks like this:

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=1842
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=isdn-test
devices=2

And the relevant bit in extensions.conf looks like this:

[isdn-test]
exten = s,1,Dial(Zap/7)

Many thanks,
Thomas
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI/Dialing out

2005-04-03 Thread Philip Hofstetter
Hi,
Philip Hofstetter wrote:
Now may next step has been to enable dialing out with the softphones.
This does not work as expected.
I was able to fix this problems by downgrading from kernel 2.6.11 to 
2.6.10. There must be a CAPI-Problem hidden somewhere.

Last saturday was so much fun for me, trying out all the stuff that can 
be done with asterisk.

Thanks to all for this wonderful program!
Philip
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI call fails

2005-03-31 Thread Andreas Meyer
Hi!

Can someone help me with a problem I have with CAPI and dialing out
or in? Installed is a B1ISA from AVM.

I have installed chan_capi-0.3.5. In modem.conf I have this entries:
[interfaces]
driver=chan_capi
type=autodetect
dialtype=tone
mode=immediate
msn=144673
device = /dev/ttyI0
device = /dev/ttyI1

in modules.conf:
[modules]
...
load = chan_capi.so
[global]
chan_capi.so=yes

in capi.conf:
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]

msn=144673
incomingmsn=*
outgoingmsn=144673
controller=1
softdtmf=1
accountcode=
context=ausgehend
echosquelch=1
echocancel=yes
echotail=64
;callgroup=1
deflect=144673
devices=2

and in extensions.conf:
exten = _3X.,1,Dial,CAPI/144673:${EXTEN:1}

with asterisk -r I get:

Connected to Asterisk 1.0.7 currently running on delta (pid = 1213)
Verbosity is at least 5
delta*CLI capi debug
CAPI Debugging Enabled
-- parse_srv: SRV mapped to host proxy.de.sipgate.net, port 5060
-- Registered SIP 'andreas' at 192.168.1.3 port 5060 expires 1800
-- Saved useragent X-Lite release 1105d for peer andreas
-- Executing Dial(SIP/andreas-7ed4, CAPI/144673:0634187482) in new stack
-- data = 144673:0634187482
-- capi request omsn = 144673
  == found capi with omsn = 144673
  == CAPI Call CAPI[contr1/144673]/3 -- creating pipe for PLCI=-1
-- Called 144673:0634187482
-- CONNECT_CONF ID=002 #0x0007 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

-- CONNECT_CONF ID=002 #0x0007 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

  == received CONNECT_CONF PLCI = 0x101 INFO = 0
-- DISCONNECT_IND ID=002 #0x0005 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Reason  = 0x3302

  == DISCONNECT_IND PLCI=0x101 REASON=0x3302
-- CAPI Hangingup
-- removed pipe for PLCI = 0x101
  == No one is available to answer at this time
-- Timeout on SIP/andreas-7ed4
  == CDR updated on SIP/andreas-7ed4
-- Executing Hangup(SIP/andreas-7ed4, ) in new stack
  == Spawn extension (ausgehend, t, 1) exited non-zero on 'SIP/andreas-7ed4'

X-lite tells me: Call failed: 403 Forbidden

I have no clue what is going wrong.

capi info tells me this:
delta*CLI capi info
Contr1: 2 B channels total, 2 B channels free.

and cat /proc/capi/controllers/1:
name b1isa-340
io   0x340
irq  7
type B1 ISA
ver_driver   3.11-03
ver_cardtype B1
ver_serial   02081722
protocol DSS1
linetype point to multipoint
cardname B1

I tried a lot different settings with no success. Can someone help
me with this? Is the B1 defective or is it the cable?

Thanks in advance!
-- 
   Andreas Meyer
   

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI call fails

2005-03-31 Thread David Woodhouse
On Thu, 2005-03-31 at 10:01 +0200, Andreas Meyer wrote:
 REASON=0x3302

This means Protocol error layer 2. Are you able to make outgoing calls
any other way using this card? Do you see anything relevant in 'dmesg'
when you make outgoing calls, or when incoming calls occur?

You don't need to configure modems.conf to use CAPI, btw.

-- 
dwmw2


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI call fails

2005-03-31 Thread Andreas Meyer
Hi!

David Woodhouse [EMAIL PROTECTED] wrote:

 On Thu, 2005-03-31 at 10:01 +0200, Andreas Meyer wrote:
  REASON=0x3302
 
 This means Protocol error layer 2. Are you able to make outgoing calls
 any other way using this card? Do you see anything relevant in 'dmesg'
 when you make outgoing calls, or when incoming calls occur?
 
 You don't need to configure modems.conf to use CAPI, btw.

ah, thanks!

I have this output after the machine rebooted:

...
Adding Swap: 511992k swap-space (priority 42)
CAPI-driver Rev 1.1.4.1: loaded
capifs: Rev 1.1.4.1
capi20: started up with major 68
kcapi: capi20 attached
capi20: Rev 1.1.4.2: started up with major 68 (middleware+capifs)
CSLIP: code copyright 1989 Regents of the University of California
ISDN subsystem Rev: 1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1 loaded
kcapi: capidrv attached
kcapi: appl 1 up
capidrv: Rev 1.1.4.1: loaded
b1: revision 1.1.4.1
b1isa: revision 1.1.4.1
kcapi: driver b1isa attached
kcapi: Controller 1: b1isa-340 attached
b1isa: AVM B1 ISA at i/o 0x340, irq 7, revision 255
b1isa-340: card 1 B1 ready.
b1isa-340: card 1 Protocol: DSS1
b1isa-340: card 1 Linetype: point to multipoint
b1isa-340: B1-card (3.11-03) now active
kcapi: card 1 b1isa-340 ready.
kcapi: notify up contr 1
capidrv: controller 1 up
capidrv-1: now up (2 B channels)
capidrv-1: D2 trace enabled
capi: controller 1 up
via-rhine.c:v1.10-LK1.1.19  July-12-2003  Written by Donald Becker
  http://www.scyld.com/network/via-rhine.html
PCI: Found IRQ 11 for device 00:11.0
PCI: Sharing IRQ 11 with 00:07.2
eth0: VIA VT6102 Rhine-II at 0xec00, 00:05:5d:a3:56:90, IRQ 11.
eth0: MII PHY found at address 8, status 0x782d advertising 01e1 Link 0021.
ne2k-pci.c:v1.02 10/19/2000 D. Becker/P. Gortmaker
  http://www.scyld.com/network/ne2k-pci.html
PCI: Found IRQ 12 for device 00:0f.0
eth1: RealTek RTL-8029 found at 0xe400, IRQ 12, 00:00:B4:9C:51:15.
usb.c: registered new driver usbdevfs
usb.c: registered new driver hub
usb-uhci.c: $Revision: 1.275 $ time 13:14:03 Feb 15 2005
usb-uhci.c: High bandwidth mode enabled
PCI: Found IRQ 11 for device 00:07.2
PCI: Sharing IRQ 11 with 00:11.0
usb-uhci.c: USB UHCI at I/O 0xe000, IRQ 11
usb-uhci.c: Detected 2 ports
usb.c: new USB bus registered, assigned bus number 1
hub.c: USB hub found
hub.c: 2 ports detected
usb-uhci.c: v1.275:USB Universal Host Controller Interface driver
IPv6 v0.8 for NET4.0
IPv6 over IPv4 tunneling driver
eth0: Promiscuous mode enabled.
device eth0 entered promiscuous mode
eth0: no IPv6 routers present
eth1: no IPv6 routers present
eth0: Promiscuous mode enabled.
kcapi: appl 2 up
kcapi: appl 2 releasing(1)
kcapi: appl 2 down
kcapi: appl 2 up
capidrv-1: DISCONNECT_IND reason 0x3301 (Protocol error layer 1 (broken line or 
B-channel removed by signalling protocol)) for plci 0x101
capidrv-1: DISCONNECT_IND reason 0x3302 (Protocol error layer 2) for plci 0x101


I don't know where to start. Dialing out gives no messages in the logfile.
Dialing in on this number gives busy on the phone (analog- or ISDN-phone)
and no messages in the logfile.

If I could get another ISDN-card running with CAPI and SuSE I would try
another card, but the B1 ist the only one I got. I can't get a FritzClassic
to work with CAPI on this SuSE-Server.

I also tried sending out a SMS with yaps using the B1. I get:

delta:/var/log # yaps 01757052847 hi
Found service D1 for 01757052847
Sending following message:
01757052847  (D1, 01757052847): hi (sent by A.Meyer!)
Trying to open /dev/ttyI0 for modem standard
[Hangup]
[Send] cr
[Cmd Mdzz 200]
[Send] ATZcr
[Expect] crATZcrcrlfOK got OK
[Send] ATE144674cr
[Expect] crlfATE144674crcrlfOK got OK
Using modem standard at 38400 bps, 8n1 over /dev/ttyI0
Trying do dial 01712521001
[Send] ATD01712521001cr
[Expect] crlfATD01712521001crcrlfBUSY got BUSY
Unable to dial D1


Thank you!

-- 
   Andreas Meyer
   

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI call fails

2005-03-31 Thread bladerunner
i've got a b1 in one of my systems (but a b1rev4, which is pci). it would help 
to know

1.) what kernel version (2.4.x, 2.6.x?)
2.) output of capiinfo
3.) output of lspci
4.) output of lsmod

support for capi20  avm b1 is in 2.6 series kernels, only thing needed is 
AFAIK to load up the correct firmwires via /etc/capi.conf (don't know how it 
is named in suse though).

michael
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI call fails

2005-03-31 Thread bladerunner
i just noticed your capi.conf. i've got a working capi.conf from one of my 
customers:

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

[interfaces]

isdnmode=ptp
msn=12345
context=in-capi
incomingmsn=*
controller=1
softdtmf=1
devices=2
mode=immediate

notice the isdnmode and mode lines, you'll have to change them based on your 
line (it seems you got point to multipoint rather than point to point) to 
ptmp. also it seems the ordering of the parameters influences the behaviour 
of chan_capi (i had to set the isdnmode-parameter at the begining of the 
block).

btw, i have no entries in modem.conf. you should only need modem.conf if you 
use isdn4linux, not capi4linux.

michael.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI call fails

2005-03-31 Thread Andreas Meyer
Hi!

bladerunner [EMAIL PROTECTED] wrote:

 i just noticed your capi.conf. i've got a working capi.conf from one of my 
 customers:
 
 [general]
 nationalprefix=0
 internationalprefix=00
 rxgain=0.8
 txgain=0.8
 
 [interfaces]
 
 isdnmode=ptp
 msn=12345
 context=in-capi
 incomingmsn=*
 controller=1
 softdtmf=1
 devices=2
 mode=immediate
 
 notice the isdnmode and mode lines, you'll have to change them based on your 
 line (it seems you got point to multipoint rather than point to point) to 
 ptmp. also it seems the ordering of the parameters influences the behaviour 
 of chan_capi (i had to set the isdnmode-parameter at the begining of the 
 block).

Yes, I have to use point to multipoint but get:
ERROR[9410]: Unknown isdnmode parameter ptmp -- ignoring
 
 btw, i have no entries in modem.conf. you should only need modem.conf if you 
 use isdn4linux, not capi4linux.

ah, didn't know this. I am using i4l and added the card with YaST.

Thanks for your suggestions!
-- 
   Andreas Meyer
   
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI call fails

2005-03-31 Thread Andreas Meyer
Hi!

bladerunner [EMAIL PROTECTED] wrote:

 i've got a b1 in one of my systems (but a b1rev4, which is pci). it would 
 help 
 to know

it's an ISA-card version 2.0

 
 1.) what kernel version (2.4.x, 2.6.x?)

delta:/etc/asterisk # uname -a
Linux delta 2.4.29 #1 Tue Feb 15 11:53:57 CET 2005 i686 unknown
(selfrolled)

 2.) output of capiinfo

delta:/etc/asterisk # capiinfo
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.17-03  (49.19)
Serial Number: 0208172
BChannels: 2
Global Options: 0x0039
   internal controller supported
   DTMF supported
   Supplementary Services supported
   channel allocation supported (leased lines)
B1 protocols support: 0x401f
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
   V.110 asynconous operation with start/stop byte framing
   V.110 synconous operation with HDLC framing
   T.30 modem for fax group 3
B2 protocols support: 0x0b1b
   ISO 7776 (X.75 SLP)
   Transparent
   LAPD with Q.921 for D channel X.25 (SAPI 16)
   T.30 fro fax group 3
   ISO 7776 (X.75 SLP) with V.42bis compression
   V.120 asyncronous mode
   V.120 bit-transparent mode
B3 protocols support: 0x803f
   Transparent
   T.90NL, T.70NL, T.90
   ISO 8208 (X.25 DTE-DTE)
   X.25 DCE
   T.30 for fax group 3

  0100
  0200
  3900
  1f40
  1b0b
  3f80
       
  0101 0002   

Supplementary services support: 0x03ff
   Hold / Retrieve
   Terminal Portability
   ECT
   3PTY
   Call Forwarding
   Call Deflection
   MCID
   CCBS


 3.) output of lspci

delta:/etc/asterisk # lspci
00:00.0 Host bridge: Intel Corp. 440LX/EX - 82443LX/EX Host bridge (rev 03)
00:01.0 PCI bridge: Intel Corp. 440LX/EX - 82443LX/EX AGP bridge (rev 03)
00:07.0 ISA bridge: Intel Corp. 82371AB/EB/MB PIIX4 ISA (rev 02)
00:07.1 IDE interface: Intel Corp. 82371AB/EB/MB PIIX4 IDE (rev 01)
00:07.2 USB Controller: Intel Corp. 82371AB/EB/MB PIIX4 USB (rev 01)
00:07.3 Bridge: Intel Corp. 82371AB/EB/MB PIIX4 ACPI (rev 02)
00:0e.0 VGA compatible controller: Trident Microsystems TGUI 9440 (rev e3)
00:0f.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8029(AS)
00:10.0 SCSI storage controller: Advanced Micro Devices [AMD] 53c974 [PCscsi] 
(rev 10)
00:11.0 Ethernet controller: VIA Technologies, Inc. Ethernet Controller (rev 43)

 4.) output of lsmod

delta:/etc/asterisk # lsmod
Module  Size  Used byNot tainted
af_packet  11816   2  (autoclean)
ipv6  146496  -1  (autoclean)
joydev  6048   0  (unused)
evdev   3936   0  (unused)
input   3104   0  [joydev evdev]
st 26672   0  (autoclean) (unused)
sg 24036   0  (autoclean)
usb-uhci   21124   0  (unused)
usbcore55968   1  [usb-uhci]
ne2k-pci4576   1
83905984   0  [ne2k-pci]
via-rhine  11752   1
mii 2320   0  [via-rhine]
crc32   2816   0  [8390 via-rhine]
b1isa   3524   1
b1 17120   0  [b1isa]
capidrv24672   1
isdn  120288   0  [capidrv]
slhc4544   0  [isdn]
capi   16960   0
capifs  3552   0  [capi]
kernelcapi 29920   4  [b1isa capidrv capi]
capiutil   22400   0  [capidrv kernelcapi]
vfat9276   0  (autoclean)
fat29816   0  (autoclean) [vfat]
tmscsim29600   2
ext3   62176   6
jbd44116   6  [ext3]

 
 support for capi20  avm b1 is in 2.6 series kernels, only thing needed is 
 AFAIK to load up the correct firmwires via /etc/capi.conf (don't know how it 
 is named in suse though).

I downloaded the newest firmware and load it with capi.conf:
# card  fileproto   io  irq mem cardnr  options
b1isa   b1.t4   DSS10x340   7   -   1

I am using i4l that came with SuSE8.0. So with isdn4linux I need
the modem.conf?
So i loaded it within modules.conf and in modem.conf I changed from
;driver=chan_capi to
driver=i4l

the problem remains the same.

This is the output I get with asterisk -vvvgc :

delta:/etc/asterisk # asterisk -vvvgc
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
  == Manager registered action Ping
  == Manager registered action Events
  == Manager registered action Logoff
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Setvar
  == Manager registered action Getvar
  

Re: [Asterisk-Users] CAPI call fails

2005-03-31 Thread Andreas Meyer
bladerunner [EMAIL PROTECTED] wrote:

 i've got a b1 in one of my systems (but a b1rev4, which is pci). it would 
 help 
 to know
 
 1.) what kernel version (2.4.x, 2.6.x?)
 2.) output of capiinfo
 3.) output of lspci
 4.) output of lsmod
 
 support for capi20  avm b1 is in 2.6 series kernels, only thing needed is 
 AFAIK to load up the correct firmwires via /etc/capi.conf (don't know how it 
 is named in suse though).

hm, I wonder why my answermail does not arrive!


-- 
   Andreas Meyer
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI/Dialing out

2005-03-28 Thread Philip Hofstetter
Hi,
after having read so much about Asterisk, I went on and tried out to 
create a little sample-setup.

I'm using a Fritz Card USB with the AVM Capi Driver and two X-Lite 
Softphones.

Dialing between the softphones makes no problem.
Calling the MSN fron an external phone also works. I'm getting to the 
asterisk demo-voicebox which works flawlessly.

Now may next step has been to enable dialing out with the softphones. 
This does not work as expected.

I can dial out and the hard phone on the other end actually rings. When 
I answer it, I can hear nothing. Noting appears on the Asterisk console, 
X-Lite still talks about trying to connect.

Now if I hang up the real phone, the state remains unchanged on the side 
of Asterisk. Both the D and B1-LEDs remain on.

Only after I hang up in the Softphone, more begins to happen in the log: 
First it tells that the call was answered, then it talks about the 
hangig up-process.

This is how a call looks:
-- Executing Dial(SIP/12346-457f, CAPI/0442607572:b012669095|30) in 
new stack
-- creating pipe for PLCI=-1
sent CONNECT_REQ MN =0x4
-- Called 0442607572:b012669095
-- CAPI[contr1/0442607572]/0 answered SIP/12346-457f  ---
-- CAPI Hangingup
sent DISCONNECT_B3_REQ NCCI=0x10101
sent DISCONNECT_REQ PLCI=0x101

I've marked the interesting line.
After begining to dial, the lines until Called 044... appear. Then 
nothing happens besides the real phone actually ringing. Even if I 
answer it, nothing happens in Asterisk or in X-Lite.

Then, when I hang up in X-Lite, the rest of above lines is printed.
If I don't answer the real phone, the line marked above is not printed. 
The rest is the same.

So it's like Asterisk not getting a signal from the CAPI-layer that the 
phone on the other side was actually answered.

What do I have to tweak? Which file do you actually need to help me? 
I've included capi.conf and the relevant parts of extension.conf below 
(as copied and pasted from various tutorials out there).

I'd gladly appriciate any help.
Philip
capi.conf:
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=0442607572
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=demo
devices=2
extension.conf:
[ch-fest-netz]
exten = _0[1-9].,1,Dial(CAPI/0442607572:b${EXTEN},30)
exten = _0[1-9].,2,Hangup
[theflintstones]
include = ch-fest-netz
exten = _[123456789],1,NoOp(call for ${EXTEN})
exten = _[123456789],2,Dial(SIP/${EXTEN},60,tr)
exten = _[123456789],3,Congestion
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI/Dialing out

2005-03-28 Thread Peer Oliver Schmidt
Philip Hofstetter wrote:
capi.conf:
[..]

[interfaces]
msn=0442607572
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=demo
devices=2
extension.conf:
[ch-fest-netz]
exten = _0[1-9].,1,Dial(CAPI/0442607572:b${EXTEN},30)

Are you sure 044260xxx is your MSN? In germany the MSN is your phone 
number without the local area code.

rgds
pos
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI/Dialing out

2005-03-28 Thread Chris W
Philip Hofstetter wrote:
msn=0442607572
incomingmsn=*
There's already been a suggestion to drop your area code. That may or 
may not work in Germany as I don't know how MSNs are presented. In 
Holland I had to have

msn=201234567
Where the number would normally be quoted as 0201234567, ie dropping the 0.
This gets corrected on called id from /etc/asterisk/capi.conf's 
[general] section which reads as follows:

[general]
nationalprefix=0
internationalprefix=00
...
Dunno if this will work for you but it all works fine for me.
cw
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI- 2 Cards

2005-03-11 Thread adria vidal
Some suggestion about how detect busy channels in a installation with 2 cards (AVM Fritz)? 
Can't find info about groups in capi channels.  Need to dial out trought some of the 4 avalaible channels.
Better try it with zaphfc ?

Adrià Vidal 
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] CAPI questions

2005-03-07 Thread Damian Funnell




Hi all,

I have two questions regarding CAPI. Excuse the fact that they are
very 'newbie' in nature, but the CAPI documentation is wafer thin!

Firstly I have four BRI adapters (all trunks and controlled by CAPI) in
my * box and I would like to know whether I can group these together
for dialling out in the same way that ZAP channels can be grouped
together.

Secondly I have a problem where * doesn't seem to recognise incoming
calls when one of the B channels is in use. If someone is on the phone
to an external number, for example, then incoming calls ring (for the
caller, at least) but * doesn't seem to have any idea that the channel
is ringing.

Lastly, my capi.conf (as below) only defines one controller as this is
what we are testing with. My understanding is that the interface block
(starting with 'msn=470' and ending with 'devices=2') needs to be
repeated for each of the four BRI adapters, but with the correct MSN
for each. The documentation I have seen is ambiguous, can anyone
confirm this is correct?

Thanks in advance,
I M Newbie.


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

[interfaces]

msn=470
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=incoming
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2

-- 
FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] CAPI questions

2005-03-07 Thread Elmar Haneke
Lastly, my capi.conf (as below) only defines one controller as this is 
what we are testing with.  My understanding is that the interface block 
(starting with 'msn=470' and ending with 'devices=2') needs to be 
repeated for each of the four BRI adapters, but with the correct MSN for 
each.
If you have different MSN then you have to repeat it for each controller.
If they are on the same MSN you can enter devices=8 and 
controller=1,2,3,4 or repeat which should also work.

Elmar
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI questions

2005-03-07 Thread Damian Funnell




Thanks Elmar. I assume it is up
to the carrier to determine the MSN for each connection?

D.

FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz


Elmar Haneke wrote:

  Lastly, my capi.conf (as below) only defines
one controller as this is what we are testing with. My understanding
is that the interface block (starting with 'msn=470' and ending with
'devices=2') needs to be repeated for each of the four BRI adapters,
but with the correct MSN for each.

  
  
If you have different MSN then you have to repeat it for each
controller.
  
  
If they are on the same MSN you can enter "devices=8" and
"controller=1,2,3,4" or repeat which should also work.
  
  
Elmar
  
___
  
Asterisk-Users mailing list
  
Asterisk-Users@lists.digium.com
  
http://lists.digium.com/mailman/listinfo/asterisk-users
  
To UNSUBSCRIBE or update options visit:
  
 http://lists.digium.com/mailman/listinfo/asterisk-users
  
  
  
  



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] CAPI trunks

2005-03-07 Thread Damian Funnell




Hi all,

Can anyone help me with a CAPI problem that I am having. I've got one
BRI trunk (will have 4 when it goes into production) and when one of
the B channels is in use (i.e. there is an incoming/outgoing call in
progress) I can't get Asterisk to answer the other ringing B channel
(Asterisk doesn't even seem to know that it is ringing).

Incoming calls work fine when no channels are in use and Asterisk will
still dial out on the second channel (if the first is in use).

Thanks,
Damian.

capi.conf:

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

[interfaces]

msn=470
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=incoming
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2





-- 
FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Capi installation with Fedora Core 3 (AVM Fritz!)

2005-03-05 Thread Rubens Sanchez
Hello! I am a newbie with asterisk, I´d like to install capi on FC3, I´ve 
tried to follow a little howto 
(http://voip-info.org/wiki-Asterisk+Linux+Fedora), but it is for FC1, and 
when I do a modprobe fcpci it fails (module not found).

Please some help!!
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] capi debugging

2005-03-03 Thread Victor Alvarez



Hi,
Regarding capi debug, I don't know how 
to translate reasons like 0x3302 or infos like 0.I didn't find any 
'translator' googleing capi debugging. Do you know about any 'translator' for 
this or should I be as clever as to know what a reason 0x3302 is?

 What is this debug for if I can't interpret 
it?

Kind regards, Victor.
From capi debug:
 == CAPI Call CAPI[contr1/number]/1 -- 
creating pipe for PLCI=-1  sent 
CONNECT_REQ MN =0x5 -- CONNECT_CONF ID=002 #0x0005 
LEN=0014 
Controller/PLCI/NCCI 
= 0x101 
Info 
= 0x0

 -- CONNECT_CONF ID=002 #0x0005 
LEN=0014 
Controller/PLCI/NCCI 
= 0x101 
Info 
= 0x0

 == received CONNECT_CONF PLCI = 0x101 INFO = 
0 -- Called @number:number -- 
DISCONNECT_IND ID=002 #0x0009 LEN=0014 
Controller/PLCI/NCCI 
= 0x101 
Reason 
= 0x3302
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] capi debugging

2005-03-03 Thread Victor Alvarez




I'm going to answer myself. I don't know If 
somebody already did it because I'm using digest mode.

CAPI specification is available at http://www.capi.org/, It explains all the 
commands and associated identifiers. Now I know that reason0x3302 in 
DISCONNECT_IND means Protocol error, Layer 2. I'llcarry on with 
myresearching from here.I don't know what is the point of 
use messages like 0x3302 instead of speak a human languagebut I've found 
my 'translator'.

Victor.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Capi channel - can I route call to another channel or back to PBX and free current channel ?

2005-02-15 Thread Robert Rozman
Hi,

I have following problem. Asterisk is connected to ISDN router on BRI
interface. ISDN PBX is connected to another channel of BRI interface. Now
I'd like to route all incoming calls first to Asterisk and then if caller
wants to talk to extension on ISDN PBX then I'd like to route call to
another capi channel but free the current one.

Is this possible at all or do I need to take 2 capi channels to route calls
?

Thanks in advance,

regards,

Rob.

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Capi channel - can I route call to another channel or back to PBX and free current channel ?

2005-02-15 Thread Shaun Ewing
On Tue, 15 Feb 2005 10:45:16 +0100, Robert Rozman [EMAIL PROTECTED] wrote:
 Hi,
 
 I have following problem. Asterisk is connected to ISDN router on BRI
 interface. ISDN PBX is connected to another channel of BRI interface. Now
 I'd like to route all incoming calls first to Asterisk and then if caller
 wants to talk to extension on ISDN PBX then I'd like to route call to
 another capi channel but free the current one.
 
 Is this possible at all or do I need to take 2 capi channels to route calls
 ?

capiECT is probably what you are after.

Have a look at http://www.voip-info.org/wiki-Asterisk+CAPI+Readme

 Thanks in advance,
 
 regards,
 
 Rob.

-Shaun
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Capi channel - can I route call to another channel or back to PBX and free current channel ?

2005-02-15 Thread Peer Oliver Schmidt
Robert Rozman wrote:
I have following problem. Asterisk is connected to ISDN router on BRI
interface. ISDN PBX is connected to another channel of BRI interface. Now
I'd like to route all incoming calls first to Asterisk and then if caller
wants to talk to extension on ISDN PBX then I'd like to route call to
another capi channel but free the current one.
IIRC you can't do this. You must connect your ISDN PBX to a HFC card and 
route it thru there.

--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CAPI not installed

2005-02-15 Thread A. Peverelli
I own a ME600 EPIA Mini-ITX main board with  the latest Debian distro 
(kernel 2.6.8) with isdnutils-base, libcapi20-dev, libcapi20-2, 
isdnactivecards installed. I have a QuadBRI module by Junghanns with 
bristuff-0.2.0-RC3a (with asterisk-1.0.3, zaptel-1.0.3 and 
libpri-1.0.3), and chan_capi-0.3.5. I followed all INSTALL instructions, 
but I have some strange behaviour. All modules seems to be correctly 
installed and actives, but on /dev I find only capi20. Anyway, starting 
Asterisk, I recevive a 'CAPI not installed!'  error on chan_capi load 
and I can't find why. Anyone has some idea?

Note: Asterisk without the QuadBRI module and chan_capi is working well, 
but I have compiled it with explicit PROC=i386, because 'uname -m' 
returns i686, but the VIA processor does not support some of 686 
instructions that the Asterisk executable uses.

# lsmod  | grep capi
capidrv297480
isdn1282041capidrv
capi177280
capifs60242capi
kernelcapi466246c4,blpci,bldma,bl,capidrv,capi
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI not installed

2005-02-15 Thread Michiel van Baak
On 11:52, Tue 15 Feb 05, A. Peverelli wrote:
 
 I own a ME600 EPIA Mini-ITX main board with  the latest Debian distro 
 (kernel 2.6.8) with isdnutils-base, libcapi20-dev, libcapi20-2, 
 isdnactivecards installed. I have a QuadBRI module by Junghanns with 
 bristuff-0.2.0-RC3a (with asterisk-1.0.3, zaptel-1.0.3 and 
 libpri-1.0.3), and chan_capi-0.3.5. I followed all INSTALL instructions, 
 but I have some strange behaviour. All modules seems to be correctly 
 installed and actives, but on /dev I find only capi20. Anyway, starting 
 Asterisk, I recevive a 'CAPI not installed!'  error on chan_capi load 
 and I can't find why. Anyone has some idea?
 
 Note: Asterisk without the QuadBRI module and chan_capi is working well, 
 but I have compiled it with explicit PROC=i386, because 'uname -m' 
 returns i686, but the VIA processor does not support some of 686 
 instructions that the Asterisk executable uses.
 

Are you running asterisk as user asterisk ?
If so, you need to add this user to the dialout group.
Otherwise it won't have access to the modem.

hope this helps.
-- 
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence.

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI not installed

2005-02-15 Thread A. Peverelli

Are you running asterisk as user asterisk ?
If so, you need to add this user to the dialout group.
Otherwise it won't have access to the modem.
hope this helps.
 

I'm running asterisk with user 'root'. Asterisk user is in the dialout 
group and I try to start asterisk as user asterisk, with the same result.

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI not installed

2005-02-15 Thread Peer Oliver Schmidt
A. Peverelli wrote:
I own a ME600 EPIA Mini-ITX main board with  the latest Debian distro 
(kernel 2.6.8) with isdnutils-base, libcapi20-dev, libcapi20-2, 
isdnactivecards installed. I have a QuadBRI module by Junghanns with 
bristuff-0.2.0-RC3a (with asterisk-1.0.3, zaptel-1.0.3 and 
libpri-1.0.3), and chan_capi-0.3.5. I followed all INSTALL instructions, 
but I have some strange behaviour. All modules seems to be correctly 
installed and actives, but on /dev I find only capi20. Anyway, starting 
Asterisk, I recevive a 'CAPI not installed!'  error on chan_capi load 
and I can't find why. Anyone has some idea?
quadBRI  CAPI!!!
The quadbri cards do not use/support CAPI. If you don't have another 
CAPI capable device in your system you can't/shouldn't use CAPI (I guess 
you could use CAPI via mISDN, but what is the point?)
--
Best regards

Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] CAPI not installed

2005-02-15 Thread A. Peverelli

Peer Oliver Schmidt posde-at-theinternet.de |Asterisk/Maestro| wrote:
A. Peverelli wrote:
I own a ME600 EPIA Mini-ITX main board with  the latest Debian distro 
(kernel 2.6.8) with isdnutils-base, libcapi20-dev, libcapi20-2, 
isdnactivecards installed. I have a QuadBRI module by Junghanns with 
bristuff-0.2.0-RC3a (with asterisk-1.0.3, zaptel-1.0.3 and 
libpri-1.0.3), and chan_capi-0.3.5. I followed all INSTALL 
instructions, but I have some strange behaviour. All modules seems to 
be correctly installed and actives, but on /dev I find only capi20. 
Anyway, starting Asterisk, I recevive a 'CAPI not installed!'  error 
on chan_capi load and I can't find why. Anyone has some idea?

quadBRI  CAPI!!!
The quadbri cards do not use/support CAPI. If you don't have another 
CAPI capable device in your system you can't/shouldn't use CAPI (I 
guess you could use CAPI via mISDN, but what is the point?)

Thank-you very much!
Your answer made me understand many things!
So... the point is that I have a Linux application CAPI speaking and I 
would like to connect it with Asterisk. Another goal is to connect 
Asterisk with an ISDN PBX, so I thought that I may do both things at 
once. Now I think that I have to change some architectural parameter... 
If someone has any suggestion on that, I will really appreciate it.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


  1   2   3   >