[asterisk-users] SIP trunk and congestion handling

[Mordechay Kaganer]

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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