[Asterisk-Users] problem with outgoing callsUnabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channelcongestion)

nik600 nik600 at gmail.com
Fri Feb 17 01:56:26 MST 2006


On 2/15/06, Michael Collins <mcollins at fcnetwork.biz> wrote:
> Nik,
>
> Looks like you're making some progress.  When I first started using A at H
> I had trouble getting the outbound dialing to work.  I wasn't sure where
> to start, so what I did was skip the macros in the dial plan.  I wanted
> to play around with exactly what digits the telco wanted to see.  So I
> put a specific extension in my [default] context like this:
>
> exten => 555,1,Dial(Zap/1/5595551212)
>
> I chose a specific Zap channel and the exact digits that I wanted to
> send to the telephone company.  This helped me figure out what to dial.
>
> The other thing you can do is log on to the CLI and turn on PRI
> debugging:
>
> pri debug span 1
>
> This will cause PRI debug messages to display on the console.  It might
> take a while but you will learn to read those debug messages.  You can
> also post them to the list and we'll help you to interpret them.
>
> -MC

ok, thanks for your support, now i've enabled debug on span 1, and
i've make a new entry in extension.conf:

exten => 444,1,Dial(Zap/0/mynumber)

when i call 444 i get in the logs:

Feb 17 03:50:59 DEBUG[3607] chan_sip.c: Setting NAT on RTP to 0
Feb 17 03:50:59 DEBUG[3607] chan_sip.c: Checking SIP call limits for device 102
Feb 17 03:50:59 DEBUG[3607] chan_sip.c: build_route: Contact hop:
<sip:102 at 192.168.100.180:5060>
Feb 17 03:50:59 VERBOSE[4262] logger.c:     -- Executing
Dial("SIP/102-2079", "Zap/0/mynumber") in new stack
Feb 17 03:50:59 NOTICE[4262] app_dial.c: Unable to create channel of
type 'Zap' (cause 0 - Unknown)
Feb 17 03:50:59 VERBOSE[4262] logger.c:   == Everyone is
busy/congested at this time (1:0/0/1)
Feb 17 03:50:59 DEBUG[4262] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.

it seems that the only information it gives mi is:

 app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.

so it seems that i don't have channel for outgoing calls? how can i check it?
maybe there is another logfile more detailed?

thanks a lot for your help...



More information about the asterisk-users mailing list