[Asterisk-Users] NOTICE[507921]: app_dial.c:742 dial_exec:
Unableto create channel of type 'Zap'
Rich Adamson
radamson at routers.com
Tue Nov 30 05:20:26 MST 2004
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 <radamson at routers.com>
>
> 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 => 7062210:9211:7062210 at 192.168.7.16
> >
> > [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
> Asterisk-Users at 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 at 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
---------------End of Original Message-----------------
More information about the asterisk-users
mailing list