Summary:ASTERISK-12656: Call failed to go through, reason (8) Congestion (circuits busy)
Reporter:Jay Vaidya (vdyjay)Labels:
Date Opened:2008-08-28 08:00:48Date Closed:2011-06-07 14:08:07
Versions:Frequency of
Environment:Attachments:( 0) messages.zip
( 1) zapata.conf
( 2) zaptel.conf
Description:When launching more than 50-60 calls in the same moment, asterisk drop the calls and give the error unable to create channels of type Zap "everyone is busy congested at this time" (cause 34 - circuit channel congestion)

[Aug 28 17:47:47] NOTICE[4329] channel.c: Unable to request channel Zap/g1/9275063731
[Aug 28 17:47:47] NOTICE[4329] pbx_spool.c: Call failed to go through, reason (8) Congestion (circuits busy)  Zap/g1/9275063731


i am running

i try with
resetinterval = never/3600

up to 40-50 calles aal went fine but after that asterisk give me same error..and after 1hr or that asterisk just stop to dial out.
i can see from command "show channels verbose"
0 active channel
0 active call

then also asterisk stop calling
Comments:By: Leif Madsen (lmadsen) 2008-08-28 08:09:56

Can you please attach the zapata.conf and zaptel.conf files, along with the Asterisk console output? I'm pretty sure this is either a configuration issue, or an issue with your service provider (or you're just out of available lines).

What was the bug number that was closed originally?

By: Jay Vaidya (vdyjay) 2008-08-28 08:37:25

Thanks for quick reply

here i attached require files

after some time calls goes in to "hangup" or "circuitbusy"

please not that i am playing with
resetinterval = never/3600

By: Jay Vaidya (vdyjay) 2008-08-28 08:38:17

if require i am ready with more testing

By: Leif Madsen (lmadsen) 2008-08-28 09:52:31

Gonna need the verbose output from the Asterisk CLI as well. Add 'verbose' to the messages section of logger.conf and do a 'logger reload'.

Can you also break each of those spans out to individual groups and retest? Does it just happen once you hit a specific span, or perhaps a certain channel?

Have you contacted your PRI provider to test the lines and watch the signalling and such go across the connection?

Also, can you please attach the output of 'pri set debug span 1' and change the '1' to whatever span is failing to place the calls? You can attach a separate file for each span if you wish.

# asterisk -cvvvn | tee /tmp/console_output.txt

By: Leif Madsen (lmadsen) 2008-08-28 09:53:41

Updating information as this is not a major issue. Also fixed the topic for brevity.

By: Jay Vaidya (vdyjay) 2008-09-02 01:05:36

Thanks for help....might be this issue s due to high CPU usage and bombarding of .call file on asterisk.

Here i close issue.
Thank you

By: Leif Madsen (lmadsen) 2008-09-02 09:31:08

You may be quite correct that the issue is due to using lots of .call files. If you are going to be creating a lot of calls in a short time frame from an automation script, then you are best off to be utilizing the Asterisk Manager Interface for those files.

I have closed this issue at your request. Please find a bug marshall on #asterisk-bugs on irc.freenode.net should you feel this issue was closed in error.