B.H. While dialing out i get a lot of AMI responses like this:
Event: Hangup Privilege: call,all Channel: SIP/TRK012-000336b0 Uniqueid: S5-1376567634.218719 CallerIDNum: XXXXXXXXX CallerIDName: YYYYYYYYYY ConnectedLineNum: XXXXXXXXX ConnectedLineName: YYYYYYYYYY *Cause: 19* *Cause-txt: User alerting, no answer* Event: OriginateResponse Privilege: call,all ActionID: 249867518_255525#YD_UFOzWQx30Wm6PM3USxGE Response: Failure Channel: SIP/TRK012/YYYYYYYYYY Context: YemotDialer_Bridge Exten: s *Reason: 8* Uniqueid: <null> CallerIDNum: XXXXXXXXX CallerIDName: YYYYYYYYYY As mentioned in the previous mails, SIP response code is 480. I would expect to get reason 3 not 8. Reason 8 is confusing my dialer software so it wants to redial the number. I use Asterisk 1.8.22. Is this a bug in asterisk or is a problem with my SIP trunk provider? On Wed, Aug 14, 2013 at 9:00 AM, Mordechay Kaganer <[email protected]>wrote: > 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 >> > -- משיח NOW! Moshiach is coming very soon, prepare yourself! יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד!
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