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: [email protected] [mailto:[email protected]] 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" <[email protected]<mailto:[email protected]>> wrote: Which version of asterisk are you using ? From: [email protected]<mailto:[email protected]> [mailto:[email protected]<mailto:[email protected]>] 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. -- משיח NOW! Moshiach is coming very soon, prepare yourself! יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד! -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
