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[pimenta at inatel.br]

Hi Sammy.

Thank you again for discussing about my subject!

The objective is really to see if there is an way to deliver more than one SIP 183 message to the caller.
For my current project, there is no intention to deliver sounds to the caller. I just want let the caller knows about some IPs, video codecs and ports from each callee side. But, all about video, not sounds. So, I don't intend to deliver two or more different sounds simultaneously to the caller.
However, if the caller gets data about each callee (IP, port, video codecs, where callees listen about video), it will be possible to provide video from the caller to each callee (using RTSP), even before some call being answered. That is, early media (only video) from the caller to each callee.


You told me "If you want all 183 msgs coming from all parties to be heard by the caller then I suggest you create a conference". So, I would like to try this. Do you know where can i find a tutorial explaining how to create a conference in dialplan?

And you told me "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". I will check it to confirm.

If even after trying all of this I will fail, so I will look for a way of passing data from every callee to the caller, maybe using SIP MESSAGE, before any call being answered.

Comment, please.

Best regards.

RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979 (Brasil)
________________________________________
De: asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] em Nome de SamyGo [govoiper@gmail.com]
Enviado: segunda-feira, 13 de julho de 2015 17:43
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] RES: How to dial extensions asynchronous-sequentially ?

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"

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 ?

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.

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,
Sammy


On Mon, Jul 13, 2015 at 3:51 PM, Rodrigo Pimenta Carvalho <pimenta@inatel.br<mailto:pimenta@inatel.br>> wrote:
Hi SamyGo.

Thank you for the replay. So, let me explain it better:

I knew that I could use something like " same = n,Dial(PJSIP/6001&PJSIP/6002) ".
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.
Yes, it is some sort of ring group implementation where users are dialled and just the first one to answer will get the call.

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.
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.

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.

Any hint will be very helpful!!

Thanks a lot!


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979
________________________________________
De: asterisk-users-bounces@lists.digium.com<mailto:asterisk-users-bounces@lists.digium.com> [asterisk-users-bounces@lists.digium.com<mailto:asterisk-users-bounces@lists.digium.com>] em Nome de SamyGo [govoiper@gmail.com<mailto:govoiper@gmail.com>]
Enviado: segunda-feira, 13 de julho de 2015 16:24
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] How to dial extensions asynchronous-sequentially ?

Hi,
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 ??

Anyway here's one way of how I think you can do.

Have a context created to dial the individual user

[dial_user]
exten => _600X.,1,Dial(PJSIP/${EXTEN})
...

and in your code change it to.

same = n,Dial(local/6001@dial_user/n&local/6002@dial_user/n)
same = n,Hangup()



On Mon, Jul 13, 2015 at 2:28 PM, Rodrigo Pimenta Carvalho <pimenta@inatel.br<mailto:pimenta@inatel.br><mailto:pimenta@inatel.br<mailto:pimenta@inatel.br>>> wrote:

Hi.


I my dialplan I have :

same = n,Dial(PJSIP/6001,10)
same = n,Dial(PJSIP/6002,30)
same = n,Hangup()


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.

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) ?
What I'm asking is if it is possible to call 6001 in an asynchronous way and then call 6002 too. Is it possible?

Any hint will be very helpful!



Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200><tel:%2B55%2035%203471%209200> RAMAL 979
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[pimenta at inatel.br]

Hi Sammy.

After answering your last message (please, see my last message), I was thinking about conferences and my main objective.
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.


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.

What do you think?

Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979 (Brasil)
________________________________________
De: asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] em Nome de SamyGo [govoiper@gmail.com]
Enviado: segunda-feira, 13 de julho de 2015 17:43
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] RES: How to dial extensions asynchronous-sequentially ?

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"

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 ?

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.

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,
Sammy


On Mon, Jul 13, 2015 at 3:51 PM, Rodrigo Pimenta Carvalho <pimenta@inatel.br<mailto:pimenta@inatel.br>> wrote:
Hi SamyGo.

Thank you for the replay. So, let me explain it better:

I knew that I could use something like " same = n,Dial(PJSIP/6001&PJSIP/6002) ".
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.
Yes, it is some sort of ring group implementation where users are dialled and just the first one to answer will get the call.

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.
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.

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.

Any hint will be very helpful!!

Thanks a lot!


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979
________________________________________
De: asterisk-users-bounces@lists.digium.com<mailto:asterisk-users-bounces@lists.digium.com> [asterisk-users-bounces@lists.digium.com<mailto:asterisk-users-bounces@lists.digium.com>] em Nome de SamyGo [govoiper@gmail.com<mailto:govoiper@gmail.com>]
Enviado: segunda-feira, 13 de julho de 2015 16:24
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] How to dial extensions asynchronous-sequentially ?

Hi,
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 ??

Anyway here's one way of how I think you can do.

Have a context created to dial the individual user

[dial_user]
exten => _600X.,1,Dial(PJSIP/${EXTEN})
...

and in your code change it to.

same = n,Dial(local/6001@dial_user/n&local/6002@dial_user/n)
same = n,Hangup()



On Mon, Jul 13, 2015 at 2:28 PM, Rodrigo Pimenta Carvalho <pimenta@inatel.br<mailto:pimenta@inatel.br><mailto:pimenta@inatel.br<mailto:pimenta@inatel.br>>> wrote:

Hi.


I my dialplan I have :

same = n,Dial(PJSIP/6001,10)
same = n,Dial(PJSIP/6002,30)
same = n,Hangup()


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.

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) ?
What I'm asking is if it is possible to call 6001 in an asynchronous way and then call 6002 too. Is it possible?

Any hint will be very helpful!



Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200><tel:%2B55%2035%203471%209200> RAMAL 979
--
_____________________________________________________________________
-- 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


--
_____________________________________________________________________
-- 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

[govoiper at gmail.com]

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).
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. 







On Mon, Jul 13, 2015 at 5:32 PM, Rodrigo Pimenta Carvalho <pimenta@inatel.br (pimenta@inatel.br)> wrote: