Help Me! - Configuring Digum Wildcard E100P with Asterisk

Hi,
I am very new to Asterisk and I have a task of configuring my Asterisk server with a E1 card.

When I dial the configured number I get the following error :

Unable to create channel of type ‘Zap’ (cause 34 - Circuit/channel congestion)

and then

Timeout, but no rule ‘t’ in context ‘sip-out’

I tried my best to search for a solution my reading hundreds of forums, articles and some Asterisk books. But no luck yet. So anyone please give me some hint, I would be so glad!

Here are my extension files (the relevant parts)

=============================================
zapata.conf :

[channels]
language=uk
context=isdn
switchtype=euroisdn
signalling=pri_cpe
dchan=16
usecallerid=yes
hidecallerid=no
pridialplan=local
usecallerid=yes
hidecallerid=no

callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
rxgain=0.0

txgain=0.0
group=1
channel=>1-10

===============================================
zaptel.conf

span = 1,1,0,ccs,hdb3,crc4
bchan=1-10
dchan=16
loadzone=uk
defaultzone=uk

===============================================
extensions.conf

[default]
exten => s,1,Hangup()

[sip-in]
;exten => s,1,Answer()
;exten => s,2,Playback(hello-world)
;exten => s,3,Hangup()
;exten => 123,1,Answer( )
;exten => 123,2,Hangup( )

[sip-out]
exten => s,1,Answer()
exten => s,2,Playback(helo-world)
exten => s,3,Hangup()

exten => 22222,1,Dial(Zap/g1/456878)
exten => 101,1,Dial(Zap/g1/0)
exten => 102,1,Dial(Zap/g1/1)
exten => 103,1,Dial(Zap/g1/3)
exten => 104,1,Dial(Zap/g1/4)
exten => 555,1,Dial(Zap/g1)

exten => 1234,1,Answer()
exten => 1234,2,Playback(hello-world)
exten => 1234,3,Hangup()

exten => 124,1,Answer()
exten => 124,2,Background(invalid)
exten => 124,3,Hangup()

[isdn]
exten => s,1,Answer()
exten => s,3,Hangup()

exten => 123,1,Dial(SIP/221)

===============================================

Please go through this and give me a hint so that I can move on! I am stuck very badly. Don’t have much time to Start from the basics of Asterisk.

can you post the debug log?

Dec 6 15:36:20 NOTICE[10373] app_dial.c: Unable to create channel of type ‘Zap’ (cause 34 - Circuit/channel congestion)
Dec 6 15:44:27 NOTICE[10477] app_dial.c: Unable to create channel of type ‘Zap’ (cause 34 - Circuit/channel congestion)
Dec 6 15:44:37 WARNING[10477] pbx.c: Timeout, but no rule ‘t’ in context 'sip-out’
Dec 6 15:58:41 NOTICE[10551] app_dial.c: Unable to create channel of type ‘Zap’ (cause 34 - Circuit/channel congestion)
Dec 6 15:58:47 NOTICE[10554] app_dial.c: Unable to create channel of type ‘Zap’ (cause 34 - Circuit/channel congestion)

first off, in zaptel.conf you should define all the channels on the PRI, not just the ones you’re using.

we could really do with a proper log output for a failed call.

you could try overlapdial=yes pridialplan=unknown prilocaldialplan=unknown in zapata.conf, replacing the pridialplan setting you have.

edit: if you don’t have much time, are you sure you want to be doing this ?

Hi,
I got this work suddenly, so I was not in touch of this area! I have no option than meeting the deadline! :astonished:)

Btw, thank you for the help.

I implemented all the channels

===============================================
new zapata.conf

[channels]
language=uk
context=isdn
switchtype=euroisdn
signalling=pri_cpe
dchan=16
usecallerid=yes
hidecallerid=no
usecallerid=yes
hidecallerid=no

overlapdial=yes
pridialplan=unknown
prilocaldialplan=unknown

callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
rxgain=0.0

txgain=0.0
group=1
channel=>1-15
channel=>17-31

===============================================

new zaptel.conf

span = 1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone=uk
defaultzone=uk

==============================================

But no progress! The same error message!

What alarms do you see in zttool?
What does the PTO say about the link.
Have you tested the link with a simulator?
what does zap show channels say?
what is in dmesg with relation to the card?

Im sure there are other questions, But thats enough for now :smile:

Ian,

No I haven’t checked the link with a simulator. Is this cause 34 an ISDN error? Is it the same error given because of a problem in the E1 link?

My setting was as follows:

|| Asterisk Server — E100P ||<==link==>|| NMS Board – VoIPserver||

But now I am in deep trouble,

The Asterisk server was removed from the network so that I have to install Asterisk in a new server and configure the E100P Digium card in the new machine!

Now I am starting at the very bottom of the ladder!!!

How do we install a E100P card and configure it properly?

Please be kind to give me a hint or a link where I can find the proper information of the way I do this configuration.

Thank you so much in advance!!!

The configuration for TE100P is similar to that for TE110P.
The documentation regarding configuration can be found at the digium site.
digium.com/en/supportcenter/ … ocs/TE110P

Finally solved the problem, which was in the link
Thank you very much for all the help given by all of you guys!