Mordechay Kaganer
2013-Aug-11 15:59 UTC
[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. -- ???? NOW! Moshiach is coming very soon, prepare yourself! ??? ?????? ?????? ?????? ??? ????? ????? ???! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130811/6035897b/attachment.htm>
Which version of asterisk are you using ? From: 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. -- ???? NOW! Moshiach is coming very soon, prepare yourself! ??? ?????? ?????? ?????? ??? ????? ????? ???! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130812/a22233d3/attachment.htm>