asterisk users - Aug 2013 - How to get the original SIP result code

B.H.

Hello, i'm using AMI Originate action (with async=true) to send outgoing
calls to a SIP trunk (using asterisk-java library to connect to AMI).

The problem is that in case of failed originate, OriginateResponse event is
returning only the reason code which is sometimes not sufficient to
determine the real cause of failure. Also, there's no way to link between
the channel that was trying to dial and failed and the original Originate
request, because OriginateResponse is issued only after the failed channel
was hang up. Only successful OriginateResponse will contain the unique id
of the established channel.

Is there any way that my AMI application can get the original SIP response
of the failed Originate action?

I'm using Asterisk 1.8.22 and slightly tweaked asterisk-java (
https://blogs.reucon.com/asterisk-java/) 1.0.0.


-- 
????? ?????? ???? ???? ???? ?????? ??????? ?????????!
???? NOW!
Moshiach is coming very soon, prepare yourself!
??? ?????? ?????? ?????? ??? ????? ????? ???!
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You can use AMI Commands and run sip set debug from that you have to
capture the response code.

http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command

Regards,


On Thu, Aug 22, 2013 at 10:43 PM, Mordechay Kaganer <mkaganer at
gmail.com>wrote:
> B.H.
>
> Hello, i'm using AMI Originate action (with async=true) to send
outgoing
> calls to a SIP trunk (using asterisk-java library to connect to AMI).
>
> The problem is that in case of failed originate, OriginateResponse event
> is returning only the reason code which is sometimes not sufficient to
> determine the real cause of failure. Also, there's no way to link
between
> the channel that was trying to dial and failed and the original Originate
> request, because OriginateResponse is issued only after the failed channel
> was hang up. Only successful OriginateResponse will contain the unique id
> of the established channel.
>
> Is there any way that my AMI application can get the original SIP response
> of the failed Originate action?
>
> I'm using Asterisk 1.8.22 and slightly tweaked asterisk-java (
> https://blogs.reucon.com/asterisk-java/) 1.0.0.
>
>
> --
> ????? ?????? ???? ???? ???? ?????? ??????? ?????????!
> ???? NOW!
> Moshiach is coming very soon, prepare yourself!
> ??? ?????? ?????? ?????? ??? ????? ????? ???!
>
> --
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On 22/08/13 15:43, Mordechay Kaganer wrote:
> B.H.
>
> Hello, i'm using AMI Originate action (with async=true) to send 
> outgoing calls to a SIP trunk (using asterisk-java library to connect 
> to AMI).
>
> The problem is that in case of failed originate, OriginateResponse 
> event is returning only the reason code which is sometimes not 
> sufficient to determine the real cause of failure. Also, there's no 
> way to link between the channel that was trying to dial and failed and 
> the original Originate request, because OriginateResponse is issued 
> only after the failed channel was hang up. Only successful 
> OriginateResponse will contain the unique id of the established channel.
>
> Is there any way that my AMI application can get the original SIP 
> response of the failed Originate action?
>
> I'm using Asterisk 1.8.22 and slightly tweaked asterisk-java 
> (https://blogs.reucon.com/asterisk-java/) 1.0.0.
>
You could dial a local channel instead and have that then go on and dial 
the destination. You will then be able to retrieve the sip response 
using something like :-

[localdial]
exten => _X.,1,Set(ddi=${CUT(EXTEN,,1)})
exten => _X.,n,Set(carrier=${CUT(EXTEN,,2)})
exten => _X.,n,Set(dialtime=${EPOCH})
exten => _X.,n,Set(_MASTERCHANNEL=${CHANNEL})
exten => _X.,n,Dial(SIP/${ddi}@${carrier})
exten => 
_X.,n,Set(SIPcause=${MASTER_CHANNEL(HASH(SIP_CAUSE,${CDR(dstchannel)}))},
Responsetime=$[${EPOCH}-${dialtime}])
exten => _X.,n,Set(SIPcode=${CUT(SIPcause," ",2)})

However you will need to set "storesipcause=yes" in your sip.conf for 
this to work. It is known to have a performance hit.

A better way would be to upgrade Asterisk and use hangup handlers. The 
documentation on how to do this in Asterisk 11 is poor and often wrong. 
I have written a blog about how we set it up as we use the feature all 
the time.
http://gblades.blogspot.co.uk/2013/07/how-to-get-sip-response-code-in.html

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