Dial -Out with Asterisk Sip-circuit - busy error
Okey İ handled my problem,
The problem is the provider. he denied my request! All providers of problems that mean the trunk!
I have an asterisk 1.6 server and a trunk. i tried to call my cell phone on the trunk (provider) when i call my cell phone it gives me:
-- Executing [0506610XXXX@phone:1] NoOp("SIP/1001-0000009b", "") in new stack
-- Executing [0506610XXXX@phone:2] Dial("SIP/1001-0000009b", "SIP/312XXXXXXX
/0506610XXXX") in new stack
== Using SIP RTP CoS mark 5
-- Called 312XXXXXXX/0506610XXXX
-- SIP/3XXXXXXXX-0000009c is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [0506610XXXX@phone:3] Hangup("SIP/1001-0000009b", "") in new stack
== Spawn extension (phone, 0506610XXXX, 3) exited non-zero on 'SIP/1001-0000009b'
I have tried various things;
#sip show peers - everything is ok all registered #sip show registry - my trunk is registered ok
my sip.conf
[general]
register=>XXXXXX:XXXXXX@ipaddress/312911
[312911]
type=friend
secret=XXXXX
username=312911
host=ipaddress
insecure=invite ,port
context=aaa
[1001]
type=friend
dtmfmode=rfc2833
context=phone
host=dynamic
secret=XX
callerID="1001"<1001>
nat=yes
my extension.conf
[myphones]
exten=> _XXX.,1,NoOp()
exten=> _XXX.,n,Dial(SIP/312911/${EXTEN})
exten=> _XXX.,n,Hangup()
[incoming]
exten=>_X.,1,NoOp()
exten=>_X.,n,Dial(SIP/1001)
exten=> _X.,n,Hangup()
[internal]
exten=>_1XXX,1,Dial(SIP/${EXTEN})
exten=>_1XXX,n,Hangup()
[phone]
include=>internal
include=>myphones
[aaa]
include=>incoming
include=>myphones
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Some common reasons that will generate such an error are:
1) The provider needs registration if you do not provide registration and only created a peer.
2) The format is incorrect. Some providers require 00 as ISD and some don't. So make sure you are using the correct number format.
3) The outgoing circuit is full. This happens where there is no channel left over from the provider, or you are not allowed to create more channels.
4) Some vendors do not support multiple registry.
For more details, you need to enable "sip set debug ip <provider ip address" and then call and check each packet.
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