[asterisk-users] CAPI didn't get a frame | avoiding initial deadlock | multiple instances of Asterisk
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
[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?
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()
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()
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()
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()
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()
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 *
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 *
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 *
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 *
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 *
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 *
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 *
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 *
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
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
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
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
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
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
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
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
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Re: [Asterisk-Users] CAPI - displaying individual MSN
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/${EXTEN:1},30,tr)
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
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Re: [Asterisk-Users] CAPI - displaying individual MSN
Hi!
msn=8304490
incomingmsn=8304490
Each user has a different numer, e.g. 83044910, 83044911, 83044912 and
so on.
This number should appear on the display of the called party, but how do
I configure this?
With the above configuration the display always shows 8304490.
I've tried to change the number in the dialplan, but this doesn't change
anything:
exten = _90[23456789].,1,SetCIDNum(83044912)
exten = _90[23456789].,2,Dial(Capi/g1/${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
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Re: [Asterisk-Users] CAPI - displaying individual MSN
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
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Re: [Asterisk-Users] CAPI problem - need help
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
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
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
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
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
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
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
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Re: [Asterisk-Users] CAPI in PTP mode not answering, dial out fine
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
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[Asterisk-Users] CAPI and Caller ID name not showing.
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.
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.
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.
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.
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
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
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AW: [Asterisk-Users] CAPI on ptp with variable length digits inphonenumber: SOLUTION for EICON
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
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
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
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
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
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
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
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
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--
Michal Bielicki
http://www.aefirion.org/
http://www.asterisk.com.pl/
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Re: [Asterisk-Users] capi problem with dialout
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
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
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
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
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
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
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
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
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--
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Re: [Asterisk-Users] capi segfault when incoming call is answered
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
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
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
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Re: [Asterisk-Users] CAPI call fails
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
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
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
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
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
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
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
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
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Re: [Asterisk-Users] CAPI/Dialing out
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
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Re: [Asterisk-Users] CAPI/Dialing out
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
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
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
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
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
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!)
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
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
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 ?
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 ?
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 ?
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
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
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
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
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
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
