<div dir="ltr">If that is the case, why are you trying asterisk ? I suggest use SIP proxy like Kamailio or OpenSIPS. When Call is initiated, create different branches to different callee destination - this will place calls simultaneously to the destination sides and will let everything coming from the callee sides to the caller (multiple 100s,180, 183).<div><br><div>At that point you can extract all the info you need. Now regarding establishing a video session and sending a video message before call gets accepted is a whole new story. </div><div><br></div><div><br></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Jul 13, 2015 at 5:32 PM, Rodrigo Pimenta Carvalho <span dir="ltr"><<a href="mailto:pimenta@inatel.br" target="_blank">pimenta@inatel.br</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Sammy.<br>
<br>
After answering your last message (please, see my last message), I was thinking about conferences and my main objective.<br>
Conferences will not work well for my case, because I it will allows more than one called party answering the call. But, after one answers the call, I need cancel the others ringing callees.<br>
<br>
<br>
In this case, maybe the best thing to do is to let the called party sends a SIP MESSAGE to the caller or to the Asterisk, even before any call being answered. Then, get the message body content and handle it via Asterisk or directly in the caller.<br>
<br>
What do you think?<br>
<span class=""><br>
Best regards.<br>
<br>
<br>
RODRIGO PIMENTA CARVALHO<br>
Inatel Competence Center<br>
Software<br>
</span>Ph: <a href="tel:%2B55%2035%203471%209200" value="+553534719200">+55 35 3471 9200</a> RAMAL 979 (Brasil)<br>
<span class="">________________________________________<br>
De: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] em Nome de SamyGo [<a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a>]<br>
</span>Enviado: segunda-feira, 13 de julho de 2015 17:43<br>
<span class="">Para: Asterisk Users Mailing List - Non-Commercial Discussion<br>
</span>Assunto: Re: [asterisk-users] RES: How to dial extensions asynchronous-sequentially ?<br>
<span class=""><br>
All I can focus now is "the objective is to see if there is an way to deliver more than one SIP 183 message to the caller"<br>
<br>
6001 has a song playing in 183 and 6002 has a "service unavailable" message, do you intend to deliver both of them simultaneously to the caller? I've seen multiple 183 Session Progress messages getting delivered to caller but what is your end game ? Play all sort of messages to the caller together ?<br>
<br>
Whoever told you about Asterisk not letting 183 go to the caller with this dialstring was right. If you want all 183 msgs coming from all parties to be heard by the caller then I suggest you create a conference, and call the 6001, and 6002 as its participant. Thats the only place where I believe the audio from different channel is mixed and streamed to users.<br>
<br>
>From SIP protocol perspective even if multiple 183 Session Progress messages reach to the Caller with each message pointing to different sources, the caller's UAC should ideally pick only one of them, the latest one I believe.<br>
<br>
BR,<br>
Sammy<br>
<br>
<br>
</span><span class="">On Mon, Jul 13, 2015 at 3:51 PM, Rodrigo Pimenta Carvalho <<a href="mailto:pimenta@inatel.br">pimenta@inatel.br</a><mailto:<a href="mailto:pimenta@inatel.br">pimenta@inatel.br</a>>> wrote:<br>
Hi SamyGo.<br>
<br>
Thank you for the replay. So, let me explain it better:<br>
<br>
I knew that I could use something like " same = n,Dial(PJSIP/6001&PJSIP/6002) ".<br>
While every extension (called phones) rings and before anyone answers, SIP 183 messages will be sent to Asterisk from callees. If a called phone answer, the others will be hanged up. It is ok for me. I want to connect the caller just to the first called party that answers.<br>
Yes, it is some sort of ring group implementation where users are dialled and just the first one to answer will get the call.<br>
<br>
If I just do " same = n,Dial(PJSIP/6001) ", there will be a SIP 183 message from 6001 to the caller. The caller will really receive that SIP 183 message. In this case, Asterisk seems to work as a proxy.<br>
However, if I do " same = n,Dial(PJSIP/6001&PJSIP/6002) ", the caller will not receive those SIP 183 messages from 6001 and 6002. In this case asterisk seems to work different of a proxy, as someone told me in this list.<br>
<br>
So, if I dial 6001 and 6002, but in asynchronous and sequentially way, I will have a chance to see if the caller will receive the SIP 183 messages from 6001 and 6002. That it, the objective is to see if there is an way to deliver more than one SIP 183 message to the caller, in a kind of ring group implementation.<br>
<br>
Any hint will be very helpful!!<br>
<br>
Thanks a lot!<br>
<br>
<br>
RODRIGO PIMENTA CARVALHO<br>
Inatel Competence Center<br>
Software<br>
</span>Ph: <a href="tel:%2B55%2035%203471%209200" value="+553534719200">+55 35 3471 9200</a><tel:%2B55%2035%203471%209200> RAMAL 979<br>
________________________________________<br>
De: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>> [<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>>] em Nome de SamyGo [<a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a><mailto:<a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a>>]<br>
<span class="">Enviado: segunda-feira, 13 de julho de 2015 16:24<br>
Para: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Assunto: Re: [asterisk-users] How to dial extensions asynchronous-sequentially ?<br>
<br>
Hi,<br>
Even you achieve that, what would be the objective? Do you want to just call the user and Hangup ? or Dial two users and connect them together ? Is this some sort of ring group implementation where users are dialled and first one to answer will get the call ??<br>
<br>
Anyway here's one way of how I think you can do.<br>
<br>
Have a context created to dial the individual user<br>
<br>
[dial_user]<br>
exten => _600X.,1,Dial(PJSIP/${EXTEN})<br>
...<br>
<br>
and in your code change it to.<br>
<br>
same = n,Dial(local/6001@dial_user/n&local/6002@dial_user/n)<br>
same = n,Hangup()<br>
<br>
<br>
<br>
</span><span class="">On Mon, Jul 13, 2015 at 2:28 PM, Rodrigo Pimenta Carvalho <<a href="mailto:pimenta@inatel.br">pimenta@inatel.br</a><mailto:<a href="mailto:pimenta@inatel.br">pimenta@inatel.br</a>><mailto:<a href="mailto:pimenta@inatel.br">pimenta@inatel.br</a><mailto:<a href="mailto:pimenta@inatel.br">pimenta@inatel.br</a>>>> wrote:<br>
<br>
Hi.<br>
<br>
<br>
I my dialplan I have :<br>
<br>
same = n,Dial(PJSIP/6001,10)<br>
same = n,Dial(PJSIP/6002,30)<br>
same = n,Hangup()<br>
<br>
<br>
The extension 6002 will not be invited until the called party 6001 hangs up or until 10 seconds if nobody answers the call in 6001.<br>
<br>
How to call 6001 and immediately call 6002, having 2 phones ringing at same time, but without doing something like this : same = n,Dial(PJSIP/6001&PJSIP/6002) ?<br>
What I'm asking is if it is possible to call 6001 in an asynchronous way and then call 6002 too. Is it possible?<br>
<br>
Any hint will be very helpful!<br>
<br>
<br>
<br>
Best regards.<br>
<br>
<br>
<br>
RODRIGO PIMENTA CARVALHO<br>
Inatel Competence Center<br>
Software<br>
</span>Ph: <a href="tel:%2B55%2035%203471%209200" value="+553534719200">+55 35 3471 9200</a><tel:%2B55%2035%203471%209200><tel:%2B55%2035%203471%209200> RAMAL 979<br>
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</div></div></blockquote></div><br></div>