Answer below. On Thu, Nov 30, 2017 at 6:46 PM, Yasuhiko Kamata <yasuhiko.kam...@nxtg.co.jp> wrote: > Hello asterisk-dev list, > > We have created a patch for use in 3PCC applications. > With this patch, asterisk can let the certain SIP phone answer or hold > through AMI action. > > [summary] > A patch for sending in-dialog SIP NOTIFY message > with "SIPnotify" AMI action for latest (14.x and 15.x) asterisk. > > [detail] > asterisk has an AMI action called "SIPnotify" in order to send any SIP > NOTIFY message. We want to let the SIP phone answer or hold by using > this action, but some SIP phones, especially AudioCodes 4xx phone, > do not accept it. > > According to our investigation, some SIP phones do not accept SIP > NOTIFY message if it's not an in-dialog (must have the same "From:", > "Call-ID", and tags as "INVITE"), so we created this patch. > > [note] > Some additional (not default) settings may be required on the phone side. > For AudioCodes' 4xx phone, these settings should be added: > > voip/talk_event/enabled=1 > voip/auto_answer/enabled=1 > voip/auto_answer_use_180/enabled=1 > > [how to use] > After applying this patch (and rebuild after that), > "SIPnotify" can be sent just like this: > > --- > Action: SIPnotify > ActionID: 3 > Channel: SIP/1001 > Variable: Event=talk > Variable: Call-ID=112233445566778899aabbccdde...@example.org:5060 > --- > > Here, channel and SIP call-ID should be changed accordingly. > SIP call-ID can be acquired by "SIPCALLID" in response of > "Status" AMI action: > > --- > Action: Status > ActionID: 2 > Channel: SIP/1001-00000001 > AllVariables: true > > Event: Status > Privilege: Call > Channel: SIP/1001-00000001 > ChannelState: 5 > ChannelStateDesc: Ringing > ... > Variable: DIALEDPEERNUMBER=1001 > Variable: SIPCALLID=112233445566778899aabbccdde...@example.org:5060 > --- > > After sending a "SIPnotify" action with "Variable: Call-ID=...", > SIP NOTIFY message will be sent to the target SIP phone with in-dialog > (i.e. the same "From:", "Call-ID", and tags as "INVITE"). > > If change "Event=talk" to "Event=hold", SIPnotify can be used to let > the phone hold (use "Event=talk" to unhold). > > Thanks, > > -- > Yasuhiko Kamata <yasuhiko.kam...@nxtg.co.jp>
Hey Yasuhiko, First off, thanks for letting us know about your interesting patch! The best way to get a patch contributed is to submit it to our code review site, gerrit.asterisk.org. From there, members of the Asterisk development team can review the code and further discuss it in a more appropriate context for code review. You can find some information about the patch contribution process at https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process Best wishes. -- Matthew Fredrickson Digium, Inc. | Asterisk Project Lead and Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
