Re: [asterisk-users] Zaptel problems on SuSE 9.3
- Original Message - From: Philipp Kempgen philipp.kemp...@amooma.de To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 05, 2009 10:39 PM Subject: Re: [asterisk-users] Zaptel problems on SuSE 9.3 Angus Asterisk schrieb: It seems that the zaptel startup script does not work. I noticed at startup the line: /etc/init.d/zaptel: line 40: /etc/init.d/functions: No such file or directory Just some feedback which might be helpful. The VIA box I am running on has an internal modem and I think that might have had a resource clash with the Digium board. So I disabled that plus other hardware devices I din't need in the bios. The zaptel startup script doesn't seem to work on suse so I added: modeprobe wctdm ztcfg -vvv asterisk in /etc/init.d/boot.local I think Suse is probably a bit over the top for asterisk. Too much stuff running. I might look at some really cut down linux distros. What do other people use? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (OT) Zaptel, SuSE 9.3, Debian
Suse 11.1 for some reason won't install on the VIA box. After installing get garbled text on screen. I want to fix this as a learning experience. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: 04 October 2009 23:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (OT) Zaptel, SuSE 9.3, Debian Just detecting this tread... Moving to Debian is quite a big step. How about updating to openSUSE_11.1 and use the prebuild asterisk packages (either zaptel or dahdi) . On the OBS they are available for 1.4.x, 1.6.0, 1.6.1 hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel problems on SUSE 9.3
Core show channeltypes: SIP Session Initiation Protocol (SIP)yes yes yes Console OSS Console Channel Driver no yes no OOH323 Objective Systems H323 Channel Driverno yes no Skinny Skinny Client Control Protocol (Skinny) no yes no Phone Standard Linux Telephony API Driver no yes no Agent Call Agent Proxy Channel yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes Local Local Proxy Channel Driver yes yes no MGCPMedia Gateway Control Protocol (MGCP)yes yes no --LI 9 channel drivers registered. Which one in above signifies the Zaptel channel? asterisk:~ # cat /proc/zaptel/* Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 (MASTER) 1 WCTDM/0/0 RED 2 WCTDM/0/1 RED 3 WCTDM/0/2 RED 4 WCTDM/0/3 RED lspci: :00:14.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I fiddled an it is working now - all green lights on on board. It seems that the zaptel startup script does not work. I noticed at startup the line: /etc/init.d/zaptel: line 40: /etc/init.d/functions: No such file or directory Line 40: # Source function library. if [ $system = redhat ]; then . $initdir/functions || exit 0 Fi The . %initdir... is line 40. Any ideas how to fix this file on suse? I think if I can fix this everything should be ok. Angus Could the system be confusing the Zaptel device for Tiger Jet? Ie loading wrong driver? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: 04 October 2009 22:47 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Zaptel problems on SUSE 9.3 On Sun, Oct 04, 2009 at 11:28:23AM +0100, Angus Asterisk wrote: Hi My asterisk output is: chan_sip.so = (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent X-Lite release 1011s stamp 41150 for peer 201 -- Executing [907768385...@default:1] Dial(SIP/201-083e75c0, ZAP/g1/907768385144|60) in new stack [Oct 4 11:54:27] WARNING[6255]: channel.c:3388 ast_request: No channel type registered for 'ZAP' Looks like chan_zap failed to load or something similar. What is the output of: (in Asterisk) core show channeltypes (In Linux) cat /proc/zaptel/* [Oct 4 11:54:27] WARNING[6255]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [907768385...@default:2] Hangup(SIP/201-083e75c0, ) in new stack == Spawn extension (default, 907768385144, 2) exited non-zero on 'SIP/201-083e75c0' when I make a call from a sip device to my outbound analog trunk using a Digium TDM card. My /etc/zaptel.conf file: loadzone=uk defaultzone=uk fxsks=1-4 I am in the uk by the way. Relevant part of /etc/astersk/zapata.conf: signalling=v23 ; added for UK CLI detection cidstart=polarity ; added for UK CLI detection context=frompstnanalog group=1 callgroup=1 pickupgroup=1 signalling=fxs_ks channel=1-4 part of extensions.conf: exten = _X.,1,Dial(ZAP/g1/${EXTEN},60) exten = _X.,2,Hangup I am running suse 9.3 on via and read article regarding old version of zaptel driver and fixed as per script - http://www.voip-info.org/wiki/view/Asterisk+Linux+SuSE So now running dmesg reveals: zaptel: unsupported module, tainting kernel. Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.12.1 Zaptel Echo Canceller: MG2 Have you actually loaded the module wctdm ? So that looks encouraging But still getting problem dialing out. Also quite worrying is that there are no lights on the Digium card. This used to work on same box and same operating system. I just can't remember how I got it to work last time. Anyone have any suggestions? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net
[asterisk-users] Zaptel problems on SUSE 9.3
Hi My asterisk output is: chan_sip.so = (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent X-Lite release 1011s stamp 41150 for peer 201 -- Executing [907768385...@default:1] Dial(SIP/201-083e75c0, ZAP/g1/907768385144|60) in new stack [Oct 4 11:54:27] WARNING[6255]: channel.c:3388 ast_request: No channel type registered for 'ZAP' [Oct 4 11:54:27] WARNING[6255]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [907768385...@default:2] Hangup(SIP/201-083e75c0, ) in new stack == Spawn extension (default, 907768385144, 2) exited non-zero on 'SIP/201-083e75c0' when I make a call from a sip device to my outbound analog trunk using a Digium TDM card. My /etc/zaptel.conf file: loadzone=uk defaultzone=uk fxsks=1-4 I am in the uk by the way. Relevant part of /etc/astersk/zapata.conf: signalling=v23 ; added for UK CLI detection cidstart=polarity ; added for UK CLI detection context=frompstnanalog group=1 callgroup=1 pickupgroup=1 signalling=fxs_ks channel=1-4 part of extensions.conf: exten = _X.,1,Dial(ZAP/g1/${EXTEN},60) exten = _X.,2,Hangup I am running suse 9.3 on via and read article regarding old version of zaptel driver and fixed as per script - http://www.voip-info.org/wiki/view/Asterisk+Linux+SuSE So now running dmesg reveals: zaptel: unsupported module, tainting kernel. Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.12.1 Zaptel Echo Canceller: MG2 So that looks encouraging But still getting problem dialing out. Also quite worrying is that there are no lights on the Digium card. This used to work on same box and same operating system. I just can't remember how I got it to work last time. Anyone have any suggestions? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel problems on SUSE 9.3
That's interesting. Do you have experience working with both SUSE and Debian? Why is Debian easier? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Josué Conti Sent: 04 October 2009 15:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zaptel problems on SUSE 9.3 Hello, you can change the O.S for Debian etch 4.0? Is more better and more easy to install the library and asterisk dependences. Please let me know Regards Josue 2009/10/4 Angus Asterisk aster...@iteloffice.com: Hi My asterisk output is: chan_sip.so = (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent X-Lite release 1011s stamp 41150 for peer 201 -- Executing [907768385...@default:1] Dial(SIP/201-083e75c0, ZAP/g1/907768385144|60) in new stack [Oct 4 11:54:27] WARNING[6255]: channel.c:3388 ast_request: No channel type registered for 'ZAP' [Oct 4 11:54:27] WARNING[6255]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [907768385...@default:2] Hangup(SIP/201-083e75c0, ) in new stack == Spawn extension (default, 907768385144, 2) exited non-zero on 'SIP/201-083e75c0' when I make a call from a sip device to my outbound analog trunk using a Digium TDM card. My /etc/zaptel.conf file: loadzone=uk defaultzone=uk fxsks=1-4 I am in the uk by the way. Relevant part of /etc/astersk/zapata.conf: signalling=v23 ; added for UK CLI detection cidstart=polarity ; added for UK CLI detection context=frompstnanalog group=1 callgroup=1 pickupgroup=1 signalling=fxs_ks channel=1-4 part of extensions.conf: exten = _X.,1,Dial(ZAP/g1/${EXTEN},60) exten = _X.,2,Hangup I am running suse 9.3 on via and read article regarding old version of zaptel driver and fixed as per script - http://www.voip-info.org/wiki/view/Asterisk+Linux+SuSE So now running dmesg reveals: zaptel: unsupported module, tainting kernel. Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.12.1 Zaptel Echo Canceller: MG2 So that looks encouraging But still getting problem dialing out. Also quite worrying is that there are no lights on the Digium card. This used to work on same box and same operating system. I just cant remember how I got it to work last time. Anyone have any suggestions? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with Digium TDM400 card
I had a working Asterisk 1.4.24.1 installation on SUSE 9 Linux but SIP only. I then downloaded and installed latest Zaptel and could not get Zaptel working. So I downloaded Asterisk again and re-installed. But still problems: Here is my ztcfg output: asterisk:/etc/asterisk # ztcfg -v Notice: Configuration file is /etc/zaptel.conf line 4: Unable to read Zaptel version information. Zaptel Version: Unknown Echo Canceller: Unknown Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels to configure. ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) My zaptel.conf file is in /etc/ fxsks=1-4 loadzone=uk defaultzone=uk I know config above is correct because used to work in an older Asterisk 1.2 installation. My zapata.conf file is in /etc/asterisk/ has this setting: signalling=fxs_ks What should I be looking at? Works ok for SIP but I want to get the analog card working. It is a TDM04B. Angus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users