[asterisk-users] SIP trunk and congestion handling

[Shishir Pokharel]

21.1.5<http://tools.ietf.org/html/rfc3261#section-21.1.5> 183 Session Progress


   The 183 (Session Progress) response is used to convey information
   about the progress of the call that is not otherwise classified.  The
   Reason-Phrase, header fields, or message body MAY be used to convey
   more details about the call progress.

21.1.2<http://tools.ietf.org/html/rfc3261#section-21.1.2> 180 Ringing





   The UA receiving the INVITE is trying to alert the user.  This

   response MAY be used to initiate local ringback.

http://tools.ietf.org/html/rfc3261#section-21.1.2

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mordechay Kaganer
Sent: Tuesday, August 13, 2013 10:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP trunk and congestion handling


B.H.

Asterisk 1.8.22

Thanks
On Aug 12, 2013 8:05 PM, "Shishir Pokharel" <Shishir.Pokharel at on24.com<mailto:Shishir.Pokharel at on24.com>> wrote:
Which version of asterisk are you using ?


From: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com> [mailto:asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] On Behalf Of Mordechay Kaganer
Sent: Sunday, August 11, 2013 8:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP trunk and congestion handling

B.H.

Hello, all. We have a dialer software that runs outgoing telephony campaigns. We have been using it successfully with PRI cards, now we're evaluating it's use also with a SIP trunk. Most of the things run perfectly good without a need to change anything except for dial string, but there's some strange problem with asterisk interpreting SIP result codes.

Our software is written in Java using asterisk-java library. It is using Asterisk's reason code from OriginateResponseEvent to determine if it should redial the number. Our consideration is that if Asterisk returns reason code 8 (Congestion) this means that the call has never actually reached the destination number, and it's OK to try to redial again.

But with SIP trunk, many times i can see a really strange sequence of events:

After INVITE i get the following responses (example from a real conversation)
[17:01:40] SIP/2.0 100 Trying
[17:01:40] SIP/2.0 183 Session Progress
[17:01:51] SIP/2.0 480 Temporarily not available

As far as i understand, this means that the remote phone was ringing for 10 seconds and then the call failed due to a timeout. As far as i understand, i'm supposed to get reason code 3, but actually the java application gets OriginateResponseEvent with failure reason code 8.

This behavior is hard to reproduce. I was trying with my own phone number and then i get the expected reason code 3, but i constantly get this situation running our customer's campaigns.


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