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 [907768385144 at 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 [907768385144 at 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? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091004/819d3f15/attachment.htm
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 <asterisk at 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 [907768385144 at 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 [907768385144 at 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 >
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 [907768385144 at 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 [907768385144 at 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: MG2Have 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.cohen at xorcom.com +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
Is inbound working? Can you see action on the CLI when you send a call to the lines attached to the card? PaulH B.Masoud @ SH wrote:> Hi > I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls > to that trunk, I am getting all circuits are busy now, do I have to do > something specific?? I am using elastix. > > Thanks. > > > > _______________________________________________ > -- 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 >
Man, thanks a lot! I just changed the name to g0 instead of DGTDM24 and it worked!! I would like to know where I can set the configuration for line tones( dial tone, call and busy tone) and where I can change different setting for polarity / current disconnect etc.. of the line? I cant find Zapata.cfg Thanks again! -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ioan Indreias Sent: Monday, October 05, 2009 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm outgoing>> DAHDI/DGTDM24/966505103250This (DGTDM24) is strange. Could you provide the setup of the DAHDI trunk? You should have something like DAHDI/g0/96.... or DAHDI/10/96.... Here are more info on this subject: http://www.mail-archive.com/asterisk-users at lists.digium.com/msg226642.html HTH, Ioan (Nini) Indreias www.modulo.ro _______________________________________________ -- 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
> > I cant find Zapata.cfgYou have a DAHDI installation thus you have to find chan_dahdi.conf. it should be located under /etc/asterisk Regarding the configuration for tones you have to check indications.conf file Best regards, Nini
Thanks, I made the zone, and now call disconnect works ok! i have one last problem, I have defined the card g0 to have 24 channels, but every time I try to call, if all ports are off the call always go to the first port, how can I balance the calls over all ports??? Any suggestions appreciated. Thanks all for the help. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ioan Indreias Sent: Monday, October 05, 2009 5:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm outgoing> > I cant find Zapata.cfgYou have a DAHDI installation thus you have to find chan_dahdi.conf. it should be located under /etc/asterisk Regarding the configuration for tones you have to check indications.conf file Best regards, Nini _______________________________________________ -- 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
B.Masoud @ SH schrieb:> I have defined the card g0 to have 24 channels, but > every time I try to call, if all ports are off the call always go to the > first port, how can I balance the calls over all ports???http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels#DialingaGroup Dial(Dahdi/r0/...) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de --
Thanks, What if I want to group a TDM2400 into 3 groups, r0/0 to r0/7 , r1/8 to r1/15 , r2/16 to r2/23 How to do that? Thanks. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Philipp Kempgen Sent: Monday, October 05, 2009 10:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm outgoing B.Masoud @ SH schrieb:> I have defined the card g0 to have 24 channels, but > every time I try to call, if all ports are off the call always go to the > first port, how can I balance the calls over all ports???http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels#DialingaGroup Dial(Dahdi/r0/...) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de -- _______________________________________________ -- 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
B.Masoud @ SH
2009-Oct-08 17:37 UTC
[asterisk-users] Asterisk debug message --- stopped sounds ???
Anyone pls???? I have seen this message " stopped sounds " while I am watching asterisk debug: -- Called 99999/0532828384 -- Call accepted by 192.168.10.220 (format ulaw) -- Format for call is ulaw>>>>-- IAX2/99999-69 stopped sounds-- IAX2/99999-69 answered SIP/xxx.xxx.xxx.xxx-b7d009a0 What does it mean?? _______________________________________________ -- 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