Single call two unique id


#1

Dear Team,

I have installed asterisk 11 and configured the voip account.And am originate the call

If the call answered am sending to context ,s,1

If the call not answering am sending to context,failed,1

Issue : When the originated call in answered am getting single entry in CDR,

But when the call is Not answered or busy am getting two CDR entry with different unique id.

Below is the command am using to originate:

Action: Originate
CallerID:+911233211233
WaitTime: 30
Channel: SIP/Voip/mynumber
Context: asteriskdial
Exten: s
Priority: 1

example : am originated call and enduser rejected the call and the call came context asteriskdial,failed,1

**But when i have NoOp the unique id in the context am getting different unique id and disposition as answered of the Busy call. and in CDR there are two entry one with actiual dial entry with proper entry and second one as wrong dispositions **

Can any one help on this??


#2

Asterisk 11 is no longer supported, so I doubt there is anyone here who would like to look into such a thing. As well Asterisk 13 has completely redone CDR support, so it may not behave the same there.


#3

Hai @jcolp

I have also tried with asterisk 13 still am getting the same .


#4

Hi,

Please check your cdr.conf file and uncomment the unanswered or set unanswered = no to avoid to create two legs for a single call.


#5

I need to execute one agi script for the unanswered and Busy calls.

I have added the extension as below

exten => failed,1,AGI(myagiscript)

or is there any option excute the agi script for unanswered call.?


#6

Use ${DIALSTATUS} and GotoIf


#7

am not making the call from dial plan to use goto if and dial status.

am originating the call.

while originating the call is there any option to get the status of failed calls …


#8

You can use local channel with the AMI to get the DIALSTATUS variable or use Getvar action

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_Getvar


#9

But in this case how can use Getvar action function.as the call is failed it will return the value??

And if dont mind can you suggest how can use the local channel instead of my sip channel?

Do u i need to make extra changes in the configuration to use local channel?


#10

https://wiki.asterisk.org/wiki/display/AST/Using+Callfiles+and+Local+Channels

(Call files are similar to originate.)


#11

You already have an answer