3

Okey İ handled my problem,

Problem is provider. it is rejected my request! All problem provider that means trunk!

I have a asterisk server 1.6 and a trunk. i tried to call my cell phone on 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 tried varios things;

#sip show peers- all ok all registered #sip show registry - my trunk is ok registered

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
Yasin Caner
  • 131
  • 1
  • 5
  • 16

2 Answers2

1

Some common causes which will generate this kind of error:

1) Provider needs registration where you are not giving register and only created peer.

2) The format is wrong. Some provider needs 00 as ISD, some do not. So check that you are using the correct number format.

3) The outbound circuit is full. It happens where there is no channel left from the provider side or you are not allowed to create more channels.

4) Some providers don't support multiple registry.

Fore more details you need to enable "sip set debug ip < provider ip address> and then make the call and check each packet.

Tommie C.
  • 12,895
  • 5
  • 82
  • 100
Anand Kumar Gupta
  • 11
  • 2
0

on asterisk console: set verbose 0 // optional set debug 0 // optional

set sip debug

it seems that your trunk is not working(wrong number format, invalid login/pass,...)

in sip messages U'll see real error

Dino Vizinger
  • 16