[Asterisk-Users] NOTICE[507921]: app_dial.c:742 dial_exec:
Unableto create channel of type 'Zap'
U. Abdullah Sheikh
ghalman at hotmail.com
Mon Nov 29 16:53:56 MST 2004
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
More information about the asterisk-users
mailing list