[Asterisk-Users] NOTICE[507921]: app_dial.c:742
dial_exec:Unableto create channel of type 'Zap'
U. Abdullah Sheikh
ghalman at hotmail.com
Wed Dec 1 08:46:01 MST 2004
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 <radamson at routers.com>
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
>
>
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