On Wed, 20 Apr 2005 18:33:44 +0000 "Jaime Blanco" <[EMAIL PROTECTED]> wrote:
Hi,

I just installed the asterisk and the X100P card. I can receive calls from PSTN and it can ring on a Grandstream SIP Phone. From the SIP Phone I can dial the demo extension on asterisk pbx. The issue is as soon as I try to dial out 92714756 or another number I received the following message:

*CLI> -- Executing Dial("SIP/1001-2b93", "Zap/g2/2714756") in new stack
Apr 20 02:27:40 NOTICE[245776]: app_dial.c:536 dial_exec: Unable to create channel of type 'Zap'
== Everyone is busy at this time
-- Executing Congestion("SIP/1001-2b93", "") in new stack
== Spawn extension (from-sip, 92714756, 2) exited non-zero on 'SIP/1001-2b93'



You are getting this because your dial plan is trying to send the connection to ZAP/g2 which is any zap channel in group number 2.


If you look in your zapata.conf below, you do not even have a group defined.

Zapata.conf is:

[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes

echocancel=yes
echocancelwhenbridged=no

rxgain=0.0
txgain=0.0

immediate=no

context=default

signalling=fxs_ks
channel=1




<SNIP>

Trying cleaning up your extensions.conf so it is a little more readable. I understand that you may just be getting started, but it is really difficult to try and decipher your extensions.conf file the way it is.
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@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

Reply via email to