Is this the only device on IRQ 12? What does ztcfg -vvv show?
Lyle ----- Original Message ----- From: "U. Abdullah Sheikh" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]>; <[EMAIL PROTECTED]> Sent: Wednesday, December 01, 2004 9:46 AM Subject: Re: [Asterisk-Users] NOTICE[507921]: app_dial.c:742 dial_exec:Unableto create channel of type 'Zap' > Hi Adamson, > > Thanks for such a comprehensive answers. Below is some more data for your > feedback. I tried all, but it is still not working. > > Any comments and advise based on below data? > > 0. The System is in Singapore. > > 1. I have an X100P Generic Clone Card bought over from eBay. > > 2. lspci output: > > 00:0e.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN > interface > Subsystem: Intel Corp.: Unknown device 0003 > Flags: bus master, medium devsel, latency 32, IRQ 12 > I/O ports at ec00 > Memory at ef001000 (32-bit, non-prefetchable) [size=4K] > Capabilities: [40] Power Management version 2 > > 3. lsmod output: > > Module Size Used by > wcfxo 12448 0 > zaptel 241028 1 wcfxo > crc_ccitt 1985 1 zaptel > > 4. /usr/sbin/zaptel/zttool output: I see the output below: > > Zaptel Tool (C)2002 Linux Support Services, Inc. > > > > âââââââââââââââââââââ⤠Zapata Telephony > Interfaces âââââââââââââââââââââââ > â > â > â Alarms Span > â > â OK Generic Clone Board 1 > â â > â > â â > > > ââââââââââââââââââ⤠Generic Clone Board 1 ââââââââââââââââââââ > â â > â â > â Current Alarms: No alarms. â > â Sync Source: Internally clocked â > â IRQ Misses: 0 â > â Bipolar Viol: 0 â > > â Tx/Rx Levels: 0/ 0 â > â Total/Conf/Act: 1/ 1/ 0 â > > > Span 1: 1 total channels, 1 configured F1=Details > F10=Quit > > > 5. the show modules from asterisk CLI ... output below: > > chan_zap.so Zapata Telephony w/PRI 0 > > > 6. Zapata config is pasted below: > > [channels] > relaxdtmf=yes > callwaiting=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > usecallerid=yes > echocancel=yes > echocancelwhenbridged=yes > rxgain=0.0 > txgain=0.0 > immediate=yes > context=bell > signalling=fxs_ks > callerid=asreceived > channel => 1 > > thanks& regards > > ----Original Message Follows---- > From: Rich Adamson <[EMAIL PROTECTED]> > > Would you tell us what country this system is in? > > The zap show channels should look something like: > phoenix*CLI> zap show channels > Chan Extension Context Language MusicOnHold > pseudo inbound-bus-x10 en default > 1 inbound-bus en default > and the 'zap show channel 1' should fill your cli screen with relevent > data. So, yes you have a problem with the zap channel, but with the > data included in your posting there isn't enough info to point to an > exact cause. > > >From the linux command line, do a 'lspci' and look for something that > says "Tiger Jet". If you don't see something related to the x100p, then > your system isn't recognizing the x100p. (I'm assuming this _is_ a > digium x100p and not one of the knockoffs.) > > >From the linux command line, do a 'cat /proc/interrupts' and look for > the x100p driver (wcfxo if memory serves correctly). Is it there? > > Change directory to /usr/src/zaptel and do a './zttool' from the > command line. Do you see the x100p listed? > > >From the linux command line, do a 'lsmod'. Is the wcfxo and zaptel > drivers listed? Does the zaptel entry have a [wcfxo] to the right > side of the line? > > >From an asterisk cli, do a 'show modules'. Do you see something like: > chan_zap.so Zapata Telephony w/PRI > > If you see acceptable entries for all of the above, then it would > appear something is very wrong with your /etc/asterisk/zapata.conf > file. Don't know what, but could be spaces inserted where there > shouldn't be, control characters embedded that can't be seen, or > whatever. Worst case, rename that file and create a new one ensuring > all entries are entered correctly. > > Rich > > ------------------------ > > Hi Rich Adamson, > > > > Thanks for your valuable reply. The telco line is connected and working > > properly. The phone number is also correct (see the debug messages). > > > > 1. I suspected it may be SIP <-> SIP issue, which might be causing SIP to > > PSTN dialout problem. > > > > 2. Is there any command, which I can use to confirm the zap channels are > > okay? > > > > 3. Also this output from Asterisk CLI is weired, would you like to > comment? > > > > > starwars*CLI> zap show channels > > > Chan Extension Context Language MusicOnHold > > > pseudo default default > > > > > > starwars*CLI> zap show channel 1 > > > Unable to find given channel 1 > > > > what should I get??? > > > > thanks & regards > > Abdullah > > > > > > ----Original Message Follows---- > > From: Rich Adamson <[EMAIL PROTECTED]> > > > > Looks like asterisk is trying to send the call out Zap/1, but is having > > an issue that appears almost like there is no telephone line attached to > > your x100p card. Is this machine located in the US and are you sure > > the pstn line is properly plugged to the card? > > Another remote possibility is that asterisk is detecting a busy signal > > on the pstn line. If you are in the US, what is 403142142? That isn't > > a standard US telephone number. (Nine digits?) Again, if this is in the > > US, best guess is that sending those digits out the pstn line is > > resulting in some sort of busy/congestion tone coming back from your > > telco. > > > > ------------------------ > > > Hi Asterisk-ians! > > > > > > Need all of your help. I am stuck with this issue for last few days. I > > have > > > one X100P installed in my system. My Asterisk is registered with > another > > > Asterisk Server/SIP provider as client and the call is successfully > > received > > > by my Asterisk server (named as starwars). > > > > > > Now, the extentions.conf file states, the incoming INTO * should go > out > > to > > > fxo as below: > > > > > > exten => s,1,Dial(Zap/1/403142142) > > > exten => s,2,Dial(Zap/1/403132663) > > > exten => s,3,hangup > > > > > > whereas other file config is as below: > > > > > > zapata.conf > > > [channels] > > > relaxdtmf=yes > > > callwaiting=yes > > > callwaitingcallerid=yes > > > threewaycalling=yes > > > transfer=yes > > > cancallforward=yes > > > usecallerid=yes > > > echocancel=yes > > > echocancelwhenbridged=yes > > > rxgain=0.0 > > > txgain=0.0 > > > immediate=yes > > > context=bell > > > signalling=fxs_ks > > > callerid=asreceived > > > channel => 1 > > > > > > zaptel > > > > > > fxsks=1 > > > loadzone=us > > > defaultzone=us > > > > > > sip.conf > > > register => 7062210:9211:[EMAIL PROTECTED] > > > > > > [MyService] > > > type=peer > > > username=7062210 > > > fromuser=7062210 > > > secret=9211 > > > host=192.168.7.16 > > > context=incoming > > > fromdomain=sipdom.inf > > > nat=no > > > canreinvite=no > > > dtmfmode=inband > > > > > > > > > so whenever the call comes in from service provider's asterisk to my > > > starwars asterisk, I get the error messages captured below: > > > > > > > > > starwars*CLI> sip show registry > > > Host Username Refresh State > > > 192.168.7.16:5060 7062210 105 Registered > > > -- Executing Dial("SIP/192.168.7.14-085a4790", "Zap/1/67742142") > in > > new > > > stack > > > Nov 30 01:41:52 NOTICE[507921]: app_dial.c:742 dial_exec: Unable to > > create > > > channel of type 'Zap' > > > == Everyone is busy/congested at this time > > > -- Executing Dial("SIP/192.168.7.14-085a4790", "Zap/1/61002663") > in > > new > > > stack > > > Nov 30 01:41:52 NOTICE[507921]: app_dial.c:742 dial_exec: Unable to > > create > > > channel of type 'Zap' > > > == Everyone is busy/congested at this time > > > -- Executing Hangup("SIP/192.168.7.14-085a4790", "") in new stack > > > == Spawn extension (incoming, s, 3) exited non-zero on > > > 'SIP/192.168.7.14-085a4790' > > > -- Executing Dial("SIP/192.168.7.14-085a4790", "Zap/1/67742142") > in > > new > > > stack > > > Nov 30 01:41:52 NOTICE[524305]: app_dial.c:742 dial_exec: Unable to > > create > > > channel of type 'Zap' > > > == Everyone is busy/congested at this time > > > -- Executing Dial("SIP/192.168.7.14-085a4790", "Zap/1/61002663") > in > > new > > > stack > > > Nov 30 01:41:52 NOTICE[524305]: app_dial.c:742 dial_exec: Unable to > > create > > > channel of type 'Zap' > > > == Everyone is busy/congested at this time > > > -- Executing Hangup("SIP/192.168.7.14-085a4790", "") in new stack > > > == Spawn extension (incoming, s, 3) exited non-zero on > > > 'SIP/192.168.7.14-085a4790' > > > > > > > > > please note the output of the following commands: > > > > > > starwars*CLI> zap show channels > > > Chan Extension Context Language MusicOnHold > > > pseudo default default > > > > > > starwars*CLI> zap show channel 1 > > > Unable to find given channel 1 > > > > > > starwars*CLI> sip show registry > > > Host Username Refresh State > > > 192.168.7.16:5060 7062210 105 Registered > > > > > > starwars*CLI> sip show peers > > > Name/username Host Dyn Nat ACL Mask Port > > > Status > > > MyService/7062210 192.168.7.16 255.255.255.255 5060 > > > Unmonitored > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > 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 > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ---------------End of Original Message----------------- > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > 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 > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users