[SOLVED] Asterisk all-circiuts-are-busy now

abakhiet · 09-12-2010, 10:36 AM

Hola,
I have a problem related to my VOIP server which is:

Code:

Centos 5.5
Asterisk 1.6
FreePBX 2.8

when i try to dial any number through my trunk link with any VOIP Provider, the server tolds me that
all circiuts are busy now, please try a gain later !!!
so, i don't knw what is happens ...
Thanks in Advance

abakhiet · 09-16-2010, 05:07 AM

please, any one give me an answer

estabroo · 09-16-2010, 07:49 AM

you need to attach to your asterisk console set the log/verbose level up and while attached make a call and see what it tells you about it. typically its an improper rule or a client that isn't connected.

abakhiet · 09-16-2010, 08:18 AM

Thanks, but i need some advice about asterisk version, i'm a bit confused use Asterisk 1.4 or 1.6 ???

estabroo · 09-17-2010, 01:24 AM

Either asterisk should work fine, by what you originally posted it looks like your using 1.6

abakhiet · 09-17-2010, 07:07 AM

excuse me, but when i troubleshoot this problem, i remove freepbx, so i delete its admin and panel directories and then i delete its Databases Asterisk and AsteriskCDR.
Then i make its installation again by:

Quote:

yum -y install freepbx

but, when it installed and i tries to login to frepbx it gives me a password, i put:

Quote:

Username: admin
Password: admin

but without any success, please help me solving it ...

abakhiet · 10-28-2010, 04:39 PM

here is the output at my cli:

Quote:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 254034983805/8300
-- SIP/254034983805-0000003f is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] NoOp("SIP/5559-0000003e", "Dial fa iled for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 0") in new stack

please help ME.

TB0ne · 10-28-2010, 06:12 PM

Quote:

Originally Posted by abakhiet

here is the output at my cli:

please help ME.

We've tried, but you don't seem to listen, which seems common in most of the threads you post. You ask for help and get it, then don't follow the instructions, or read the links you get.

Did you not read the other response you got?

Quote:

Originally Posted by estabroo

you need to attach to your asterisk console set the log/verbose level up and while attached make a call and see what it tells you about it. typically its an improper rule or a client that isn't connected.

Did you attach to your asterisk console? Did you set the verbose level higher, and watch what happens when you make a call?? Did you check the rules/clients??

Asking for help over and over again, then not following up/following through, is pointless.

abakhiet · 10-28-2010, 08:38 PM

Quote:

Originally Posted by TB0ne

We've tried, but you don't seem to listen, which seems common in most of the threads you post. You ask for help and get it, then don't follow the instructions, or read the links you get.

Did you not read the other response you got?

Did you attach to your asterisk console? Did you set the verbose level higher, and watch what happens when you make a call?? Did you check the rules/clients??

Asking for help over and over again, then not following up/following through, is pointless.

Thanks for your comment, the previous screenshot i gave you was from my asterisk console and it tells that the server is selecting the proper trunk, but it gives the trunk is congested.
I'm sorry again and again, but i think this is the BEST forum in linux services and applications as well.

estabroo · 10-29-2010, 06:59 AM

check your sip peers and see if that trunk is connected, might want to add a qualify=yes to it's sip entry