B.H. But if the final response is 480 doesn't it mean that the call was placed but there was no reply? On Aug 13, 2013 10:30 PM, "Shishir Pokharel" <[email protected]> wrote:
> *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]> > wrote:**** > > Which version of asterisk are you using ? **** > > **** > > **** > > *From:* [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 >
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