Hi Joshua Colp. Thank you very much for alerting me about the impossibility of forwarding the SIP 183 messages from callees to caller, via Asterisk, when more than 1 callee ring at same time.
In my project the caller software (a proprietary softphone) needs to know some information about the callees, while they are still all ringing. Such information will be used to create early media (only video) from caller to all callees. For example, the caller softphone should receive the IPs and ports where each callee will listen to video data. The caller softphone will use RTSP to create such early media. That is why I was investigating an way of passing SIP 183 messages from callees to the caller. However, as you told me about such impossibility, now I have to discover a way of collecting such callees' media information and deliver it to the proprietary caller software. So, I ask you: 1 - Is there a way of collecting information from SIP messages that arrives in Asterisk, in dial plan (by means of application or functions)? If yes, I could pass it to a external software. 2- Is there a way of handling SIP 183 or SIP 180 messages in dial plan and forward such messages to another destiny, as in a proxy? 3 - Should I use Asterisk REST Interface to collect information from SIP messages that pass in the current channel of a call, whether I need collect it and pass to a proprietary software? I was reading about ARI today. 4 - By the way, can an external application, using ARI, send requests to the Asterisk, even when such application is not invoked by a dial plan? That is, can an external application decide by itself to contact a Asterisk REST interface? Any hint about early media (video) with asterisk will be very helpful to me, as I'm completely beginner in this field. Thanks a lot! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 (Brasil) ________________________________________ De: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] em Nome de Joshua Colp [jc...@digium.com] Enviado: quarta-feira, 8 de julho de 2015 11:53 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] How may SIP 183 messages a caller receives when many callee rings? Rodrigo Pimenta Carvalho wrote: > Hi. > > I have a beginner conceptual question about Asterisk: > > Let's suppose that there are 4 softphones registered in my Asterisk > and all of them are currently online. In addiction , there is no > call. > > Suddenly, one of these softphones sends a SIP message to the > Asterisk. In this case the dialplan will execute the instruction ' > exten => 2005,1,Dial(SIP/2000&SIP/2001&SIP/2002, 30) ' > > All softphones (2000, 2001 and 2002) will ring. These are proprietary > softphones and all of then will reply with SIP 183 message. SIP 183 > will contain SDP with media information. > > The question is: > > Will the caller receive SIP 183 from each callee? That is, will it > receive 3 SIP 183 messages? It is important to the caller receives a > SIP 183 message from each callee, because this caller needs to send > early media (video) to every callee. > > Or, will Asterisk send just one message SIP 183 to the caller, with > some kind of generic SDP message? Asterisk isn't a proxy, so it won't forward all 3 and it won't forward media from all 3. Right now the Dial application is simple and just doesn't forward media in this scenario. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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
