Re: [asterisk-users] RES: Can I use PJSIP_HEADER to read the SIP 183 message header?
Ok. Thank you! 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 Yaron Nachum nachum.ya...@gmail.com Enviado: segunda-feira, 20 de julho de 2015 02:53 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] RES: Can I use PJSIP_HEADER to read the SIP 183 message header? One way to do it is to use Transfer. This will cause the callee to send 302 redirect to the caller. The caller then will jump to the extension specified in the contact. You will have to dial again to the callee in the new extension. This solution will increase the traffic on your asterisk and you have to be careful from loops. ??? ??? ??, 10 ? 2015 ?-21:37 ??? ?Rodrigo Pimenta Carvalho?? ?pime...@inatel.brmailto:pime...@inatel.br??:? Ok Mark Michelson. Thank you very much! You answer tells me that I was in the wrong path trying to access information from SIP 183 message. I need to find a way to let the callee pass information/data to the caller, even before accepting the call. That is, send data during the ringing time. And in my case, there will be more than one callee ringing at same time. As ASTERISK will not forward each SIP 183 message to the caller, I intend to get data from callees in dialplan by some another way before the call being accepted. 1- Is there any way to do that? 2 - SIP MESSAGE, if sent by the calle, enters the dialplan? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9300 (Brasil) De: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] em Nome de Mark Michelson [mmichel...@digium.commailto:mmichel...@digium.com] Enviado: sexta-feira, 10 de julho de 2015 15:14 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] Can I use PJSIP_HEADER to read the SIP 183 message header? On 07/10/2015 11:53 AM, Rodrigo Pimenta Carvalho wrote: Hi. The ASTERISK wiki has a page showing the function PJSIP_HEADER(). However, it doesn't explain if such function works only over SIP INVITE messages or if it can be use, for example, to read headers from others types of SIP messages too. So, can I use PJSIP_HEADER to read the SIP 183 message header? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 (Brasil) Unfortunately, PJSIP_HEADER() cannot be used on responses because SIP responses do not enter the dialplan. -- _ -- 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
Re: [asterisk-users] How to dial extensions asynchronous-sequentially ?
Hi Pete.
No problem!
Maybe I will use only OpenSIPS, because it may be enough for me. But I still
have to investigate some points.
As I was learning the past few days, due to the fact that Asterisk is not a SIP
Proxy, it might cause some more difficult in my project.
Best regards.
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 Pete Mundy
p...@fiberphone.co.nz
Enviado: quarta-feira, 15 de julho de 2015 18:35
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] How to dial extensions asynchronous-sequentially ?
Heya Rodrigo
My apologies for the misunderstanding re the delay.
I think the 183 messages problem stems from Asterisk being a B2BUA not a proxy
and therefore not the tool or this job. But others have more skill around that
area than I do so please confirm that before accepting it as fact!
Hope you get it resolved. Sorry to muddy the waters :)
Pete
On 16/07/2015, at 9:24 AM, Rodrigo Pimenta Carvalho
pime...@inatel.brmailto:pime...@inatel.br wrote:
Hi Sammy and Pete.
Sammy, you are correct. But your example doesn't allow Asterisk forward every
SIP 183 message to the caller.
Pete, in fact, I'm not looking for a delayed ring. All extensions must ring at
same time. I got a kind of solution by using:
exten = _6XXX,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})}, 60)
However, the Asterisk is rewriting the SDP content of SIP 183 messages, before
forwarding it to the caller. That is the new question I would like to solve,
because in my project the caller must receive the SIP 183 from callee as it was
originally wrote.
Thanks and regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979 (Brasil)
De:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
em nome de Pete Mundy p...@fiberphone.co.nzmailto:p...@fiberphone.co.nz
Enviado: quarta-feira, 15 de julho de 2015 18:16
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] How to dial extensions asynchronous-sequentially ?
Heya Rodrigo
Not sure, but this expansion on Sammy's concept may help you achieve the
delayed ring on the secondary extensions you were looking for.
exten = _600.,1,Dial(PJSIP/${EXTEN})
exten = _600.,n,Hangup
exten = _600.wait5,1,Wait(5)
exten = _600.wait5,n,Dial(PJSIP/${EXTEN:0:4})
exten = _600.wait5,n,Hangup
exten = 555,1,Dial(LOCAL/6001LOCAL/6002.wait5)
exten = 555,n,Hangup
So you dial '555' and it rings 6001, then 5 second later (assuming 6001 isn't
answered yet) 6002 starts ringing too (first to answer gets it).
Pete
On 14/07/2015, at 7:24 AM, SamyGo
govoi...@gmail.commailto:govoi...@gmail.com wrote:
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/nlocal/6002@dial_user/n)
same = n,Hangup()
--
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[asterisk-users] How to create direct media with PJSIP.conf configurations in Asterisk 13?
Dear Asterisk-Users, By means of Asterisk 11 and sip.conf, I got success implementing early media. That is, all information that come from callee (SIP 183 message/ SDP) is passed to the caller without any modification in the SDP body. However, in Asterisk 13 and using pjsip.conf I'm still failing to do the same thing. See: Softphojne1 Asterisk ---Softphone2 | --SIP INVITE | | SIP INVITE| . . . . . . . . |--SIP 183 -| SDP : Media Description, name and address (m): audio 4000 RTP/AVP 8 96 Media Description, name and address (m): video 5000 RTP/AVP 97 |--SIP 183-| SDP :Media Description, name and address (m): audio 13258 RTP/AVP 8 96 Media Description, name and address (m): video 16002 RTP/AVP 97 So, there is bit modification in SDP body, caused by Asterisk. As long as I'm intending to implement direct media, I believe that Asterisk 13 has some special configuration to be done in PJSIP.conf file, that will allow things work very well again, as in Asterisk 11 and sip.conf. How to configure pjsip.con file or Asterisk, to run direct-media? Or , where to find a tutorial about it on Internet? Any hint will be very helpful! Thanks a lot. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAl 979 (Brasil) -- _ -- 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
Re: [asterisk-users] How to create direct media with PJSIP.conf configurations in Asterisk 13?
Thank you Joshua! In this case I finally decide to use SIP Proxy. I have to start testing the SIP Proxy today. Best Regards. 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: quinta-feira, 16 de julho de 2015 10:54:56 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] How to create direct media with PJSIP.conf configurations in Asterisk 13? Rodrigo Pimenta Carvalho wrote: Dear Asterisk-Users, By means of Asterisk 11 and sip.conf, I got success implementing early media. That is, all information that come from callee (SIP 183 message/ SDP) is passed to the caller without any modification in the SDP body. PJSIP does not support early direct media. It is only once the call has been established that it will be done. -- 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
[asterisk-users] How to call a group of peers all registered with the same login?
Dear Asterisk-Users, We have one Asterisk end N softphones that will be registered on it. We need to configure the Asterisk to get the following scenario working well: Some softphones (N -1) is to be registered using the same login 6001. So, there will be N-1 users 6001 registered in the SIP REGISTRAR (Asterisk). One softphone is to be registered with the login 6000. The softphone 6000 must call the others N-1 peers. So, the sofphone 6000 will dial 6001 and all the others N-1 peers must ring. For that, I suspect that some special configuration must be done in the pjsip.conf file, for the N users. But, what exactly must be configured in Asterisk to make it run well? By the way, is there some special dial plan extension to be used? Just to use Dial(SIP/6001) is not enough. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 (Brasil) -- _ -- 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
Re: [asterisk-users] How to dial extensions asynchronous-sequentially ?
Hi Sammy and Pete.
Sammy, you are correct. But your example doesn't allow Asterisk forward every
SIP 183 message to the caller.
Pete, in fact, I'm not looking for a delayed ring. All extensions must ring at
same time. I got a kind of solution by using:
exten = _6XXX,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})}, 60)
However, the Asterisk is rewriting the SDP content of SIP 183 messages, before
forwarding it to the caller. That is the new question I would like to solve,
because in my project the caller must receive the SIP 183 from callee as it was
originally wrote.
Thanks and regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979 (Brasil)
De: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com em nome de Pete Mundy
p...@fiberphone.co.nz
Enviado: quarta-feira, 15 de julho de 2015 18:16
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] How to dial extensions asynchronous-sequentially ?
Heya Rodrigo
Not sure, but this expansion on Sammy's concept may help you achieve the
delayed ring on the secondary extensions you were looking for.
exten = _600.,1,Dial(PJSIP/${EXTEN})
exten = _600.,n,Hangup
exten = _600.wait5,1,Wait(5)
exten = _600.wait5,n,Dial(PJSIP/${EXTEN:0:4})
exten = _600.wait5,n,Hangup
exten = 555,1,Dial(LOCAL/6001LOCAL/6002.wait5)
exten = 555,n,Hangup
So you dial '555' and it rings 6001, then 5 second later (assuming 6001 isn't
answered yet) 6002 starts ringing too (first to answer gets it).
Pete
On 14/07/2015, at 7:24 AM, SamyGo
govoi...@gmail.commailto:govoi...@gmail.com wrote:
Anyway here's one way of how I think you can do.
Have a context created to dial the individual user
[dial_user]
exten = _600X.,1,Dial(PJSIP/${EXTEN})
...
and in your code change it to.
same = n,Dial(local/6001@dial_user/nlocal/6002@dial_user/n)
same = n,Hangup()
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] RES: RES: RES: How to dial extensions asynchronous-sequentially ?
Hi Sammy. Thank you very much for you help! Answering your questions: a. I don't know almost anything about SIP Proxies. I just knew about Asterisk. But, I will investigate about SIP Proxies today, thankful your citation. b. We have already built a system that uses Asterisk, RTSP and provide early media (only video) very well. But, for such system we don't have a kind of ring group implementation where users are dialed and first one to answer will get the call. b. According to some simple requirements from my current project, I thought that Asterisk could be the best choice. Let me ask you (about the requirements of my project): 1 - Can those SIP proxies take care of SIP REGISTER messages? Is it possible to a peer to register itself in a SIP Proxy? 2 - Does a SIP proxy have some kind of script, as an Asterisk dialplan, to let the programmer handle calls and SIP messages, even in a simpler way? 3- Does a SIP proxy provide a way of recording information about SIP messages into a Database? 4 - Can those SIP proxies communicate with a peer (softphone for example) using some security schema (TLS, SSL, etc)? I guess these are my last questions about the case and then I will go ahead on my own. Any hint will be very helpful ! Best Regards. 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 SamyGo [govoi...@gmail.com] Enviado: segunda-feira, 13 de julho de 2015 18:57 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] RES: RES: How to dial extensions asynchronous-sequentially ? 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 pime...@inatel.brmailto:pime...@inatel.br wrote: 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 9200tel:%2B55%2035%203471%209200 RAMAL 979(Brasil) De: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] em Nome de SamyGo [govoi...@gmail.commailto:govoi...@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 pime...@inatel.brmailto:pime...@inatel.brmailto:pime...@inatel.brmailto:pime...@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/6001PJSIP/6002) . While every extension (called phones
[asterisk-users] How to dial extensions asynchronous-sequentially ?
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/6001PJSIP/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 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
[asterisk-users] RES: How to dial extensions asynchronous-sequentially ?
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/6001PJSIP/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/6001PJSIP/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 RAMAL 979
De: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] em Nome de SamyGo [govoi...@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/nlocal/6002@dial_user/n)
same = n,Hangup()
On Mon, Jul 13, 2015 at 2:28 PM, Rodrigo Pimenta Carvalho
pime...@inatel.brmailto:pime...@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/6001PJSIP/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 9200tel:%2B55%2035%203471%209200 RAMAL 979
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[asterisk-users] RES: RES: How to dial extensions asynchronous-sequentially ?
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-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] em Nome de SamyGo [govoi...@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 pime...@inatel.brmailto:pime...@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/6001PJSIP/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/6001PJSIP/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 9200tel:%2B55%2035%203471%209200 RAMAL 979 De: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] em Nome de SamyGo [govoi...@gmail.commailto:govoi...@gmail.com] Enviado: segunda-feira, 13 de julho de 2015 16:24 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto
[asterisk-users] RES: RES: How to dial extensions asynchronous-sequentially ?
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-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] em Nome de SamyGo [govoi...@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
pime...@inatel.brmailto:pime...@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/6001PJSIP/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/6001PJSIP/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 9200tel:%2B55%2035%203471%209200 RAMAL 979
De:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
em Nome de SamyGo [govoi...@gmail.commailto:govoi...@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/nlocal/6002@dial_user/n)
same = n,Hangup()
On Mon, Jul 13, 2015 at 2:28 PM, Rodrigo Pimenta Carvalho
pime...@inatel.brmailto:pime...@inatel.brmailto:pime...@inatel.brmailto:pime...@inatel.br
wrote:
Hi.
I my dialplan I have :
same = n,Dial(PJSIP/6001,10)
same = n,Dial(PJSIP/6002,30)
same = n,Hangup
[asterisk-users] RES: Can I use PJSIP_HEADER to read the SIP 183 message header?
Ok Mark Michelson. Thank you very much! You answer tells me that I was in the wrong path trying to access information from SIP 183 message. I need to find a way to let the callee pass information/data to the caller, even before accepting the call. That is, send data during the ringing time. And in my case, there will be more than one callee ringing at same time. As ASTERISK will not forward each SIP 183 message to the caller, I intend to get data from callees in dialplan by some another way before the call being accepted. 1- Is there any way to do that? 2 - SIP MESSAGE, if sent by the calle, enters the dialplan? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9300 (Brasil) De: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] em Nome de Mark Michelson [mmichel...@digium.com] Enviado: sexta-feira, 10 de julho de 2015 15:14 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] Can I use PJSIP_HEADER to read the SIP 183 message header? On 07/10/2015 11:53 AM, Rodrigo Pimenta Carvalho wrote: Hi. The ASTERISK wiki has a page showing the function PJSIP_HEADER(). However, it doesn't explain if such function works only over SIP INVITE messages or if it can be use, for example, to read headers from others types of SIP messages too. So, can I use PJSIP_HEADER to read the SIP 183 message header? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 (Brasil) Unfortunately, PJSIP_HEADER() cannot be used on responses because SIP responses do not enter the dialplan. -- _ -- 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
[asterisk-users] RES: Messages out of calls. Is it really possible?
Hi Matthew Jordan Thank you very very much! Now it seems to me that I have a direction to follow! My intention is to create a way of receiving data from callees, in asterisk, even before the call being accepted by one of them. In my project there will be more than one callee ringing at same time. In my project, when more than one callee rings, all of them sends SIP 183 message to asterisk. However, as long as asterisk doesn't forward every SIP 183 message to the caller, I have to find a way to callees send some data to the asterisk, containing information about media, for example. In asterisk I intend do collect those information and pass it to the caller, to work around those not forwarded SIP 183 messages. If it can work, I will try to implement early media with video. Can you comment about my idea? Do you think it sounds feasible? Best regards!! 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 Matthew Jordan [mjor...@digium.com] Enviado: sexta-feira, 10 de julho de 2015 15:29 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] Messages out of calls. Is it really possible? On Fri, Jul 10, 2015 at 11:51 AM, Rodrigo Pimenta Carvalho pime...@inatel.br wrote: Hi. I have read in some web sites that ASTERISK can support messages out of calls. What does it exactly means? 1 - Can a dialplan script accept and handle a message from a callee party, even before the call be connected? Since it is out of call, yes. SIP MESSAGE requests are handled by the respective channel driver (chan_sip or the res_pjsip stack) and passed to the dialplan using a special hidden channel, Message. That channel caries the payload and some meta information about the MESSAGE request, which can be accessed using the generic out-of-call messaging functions [1]. Likewise, you can send an out of call SIP MESSAGE request using MessageSend [2]. Note that all of this has been supported since Asterisk 10. 2 - Can a ringing callee send SIP MESSAGE to the ASTERISK even before answer the call? Yes, hence the term out-of-call. 3- Could I use dialplan function MESSAGE() to receive SIP messages from callees, even before the call be connected? It does not receive messages; it accesses data on the message currently being serviced by the executing Message channel. chan_sip/res_pjsip will receive and dispatch MESSAGE requests at any point in time. They have nothing to do with your normal SIP or PJSIP channels, and hence nothing to do with whatever INVITE request derived channels are currently executing in the dialplan. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_MESSAGE and https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_MESSAGE_DATA [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MessageSend -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
[asterisk-users] Messages out of calls. Is it really possible?
Hi. I have read in some web sites that ASTERISK can support messages out of calls. What does it exactly means? 1 - Can a dialplan script accept and handle a message from a callee party, even before the call be connected? 2 - Can a ringing callee send SIP MESSAGE to the ASTERISK even before answer the call? 3- Could I use dialplan function MESSAGE() to receive SIP messages from callees, even before the call be connected? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979(Brasil) -- _ -- 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
[asterisk-users] What Dial Plan function can access the contents of the SDP ?
Dear ASTERISK-users, What Dial Plan function can access the contents of the SDP ? If there is no Dial Plan Function for that, is there some another way to access contents of the SDP? Maybe via ARI ou AGI? If there is, how to access the SDP that comes with the SIP 183 response? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 (Brasil) -- _ -- 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
[asterisk-users] RES: How many SIP 183 messages a caller receives when many callee rings?
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/2000SIP/2001SIP/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
[asterisk-users] RES: Fwd: What database should I use, for simple data storing? SQLite or the buitin one?
Hi. Thank you for your instruction! What I need is simplicity. That is, a simple solution (no relational data base) will fit very well now. In this case I will start investigating about how to use the Asterisk (version 13 or later) builtin database. Is it SQLite? How to access it via dial plan, etc? What must I configure in my asterisk? What must i install to use the builtin database? Where to find a tutorial with explanations about such questions? Best regards. 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 Technical Support [supp...@telium.ca] Enviado: terça-feira, 7 de julho de 2015 11:51 Para: asterisk-users@lists.digium.com Assunto: Re: [asterisk-users] Fwd: What database should I use, for simple data storing? SQLite or the buitin one? To some extent the answer depends on how you want to use it overall, and what you already have installed. We did something similar on a project where we created a simple app accessible via AGI, and it stored/retrieved data to/from anXML file. If your access frequency is low enough that might be a good solution. On the other hand if you need complex query capability you should stay on the SQL side. If you already have MySQL installed for other Asterisk features (eg: CDR, or if you use FreePBX) then you might as well use that. -- _ -- 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
[asterisk-users] RES: What database should I use, for simple data storing? SQLite or the buitin one?
Hi Antony. Thank you for your replay. I have decided to use the builtin database, according to others help that I have kindly received in this discussion list. What I need is a simple solution, not a relational database one. Best regards. 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 Антон Сацкий [satski...@gmail.com] Enviado: terça-feira, 7 de julho de 2015 11:32 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] What database should I use, for simple data storing? SQLite or the buitin one? Propose U to use Mysql 2015-07-07 17:26 GMT+03:00 Rodrigo Pimenta Carvalho pime...@inatel.brmailto:pime...@inatel.br: Hi. I was studying about how to use databases in Asterisk, accessing it from the dial plan. In my project, my dial plan will have to store simple data (ex: IP number, port number, device name, etc) in a persistent way, so that it will be possible to retrieve such information in future moments, still via dial plan. For this case, I would like to know? 1. What is the best choice for storing and retrieving simple data , with dial plan instructions: SQLite or the builtin database option? Consider that I'm worried about installation, configuration and use difficulties. 2. Does Asterisk 13 come with SQLite ready for use or have I to install this database separately and configure it to be accessible in dial plan? 3. Where can I find tutorials about using SQLite or the builtin database for storing simple that? P.S.: I'm not interested in storing CDR data. Any hint will be very helpful! Thanks a lot! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200tel:%2B55%2035%203471%209200 RAMAL 979(Brasil) -- _ -- 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 -- Best regards Antony моб (066) 919-75-33 моб (063) 656-43-40 satski...@gmail.commailto:mail%3asatski...@gmail.com -- _ -- 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
[asterisk-users] RES: What database should I use, for simple data storing? SQLite or the buitin one?
Hi John. Thank you very much for you reply. It is exactly what I was needing to know. Now I will study about the use of SQLite + Asterisk, because MySQL will not be necessary in my solution. A relational database will not be necessary. I 'm needing simplicity. Do you know where can I find a tutorial about accessing and using the asterisk builtin database, considering Asterisk 13 or later? Any hint will be very helpful. Thanks 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 Tech Support [aster...@voipbusiness.us] Enviado: terça-feira, 7 de julho de 2015 11:58 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Assunto: Re: [asterisk-users] What database should I use, for simple data storing? SQLite or the buitin one? I believe that Asterisk 1.8 and older uses the BerkeleyDB for Asterisk's internal database (AKA the Astdb) and in newer versions use SQLite. However, the basic functionality is the same. Whether you use the Astdb or MySQL really depends on what you want to do with it. The AstDB is not a relational database like MySQL, it simply a key/value store. If you can get away with that, and you need simplicity, then the AstDB is the way to go. If you need MySQL you'll probably end up having to write AGI scripts to access it. Like I said, it all depends on what your needs are. Regards; John -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo Pimenta Carvalho Sent: Tuesday, July 07, 2015 10:26 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] What database should I use, for simple data storing? SQLite or the buitin one? Hi. I was studying about how to use databases in Asterisk, accessing it from the dial plan. In my project, my dial plan will have to store simple data (ex: IP number, port number, device name, etc) in a persistent way, so that it will be possible to retrieve such information in future moments, still via dial plan. For this case, I would like to know? 1. What is the best choice for storing and retrieving simple data , with dial plan instructions: SQLite or the builtin database option? Consider that I'm worried about installation, configuration and use difficulties. 2. Does Asterisk 13 come with SQLite ready for use or have I to install this database separately and configure it to be accessible in dial plan? 3. Where can I find tutorials about using SQLite or the builtin database for storing simple that? P.S.: I'm not interested in storing CDR data. Any hint will be very helpful! Thanks a lot! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979(Brasil) -- _ -- 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
[asterisk-users] Can I use ARI to update the builtin database, without executing the dial plan?
Hi. In my dial plan I can use the following commands to access and handle data from the builtin database. DB DB_DELETE DB_EXISTS DB_KEYS It is OK for me. However, in my current project there will be an application responsible for recording new information in the builtin database. So, I need to know: 1. Is it possible to access the builtin database, by means of ARI ? 2. If it is possible by ARI, where can I find a tutorial about it? 3. Can I do something like this?: My APP --sends data to a REST service Asterisk REST Interfacethe data is put into the builtin database The builtin database That is, can I send data to the builtin database, using REST interfaces, but without executing any dial plan? If yes, my app will be able to update data without executing the dial plan. Any hint will be very helpful! Best Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979(Brasil) -- _ -- 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
[asterisk-users] What database should I use, for simple data storing? SQLite or the buitin one?
Hi. I was studying about how to use databases in Asterisk, accessing it from the dial plan. In my project, my dial plan will have to store simple data (ex: IP number, port number, device name, etc) in a persistent way, so that it will be possible to retrieve such information in future moments, still via dial plan. For this case, I would like to know? 1. What is the best choice for storing and retrieving simple data , with dial plan instructions: SQLite or the builtin database option? Consider that I'm worried about installation, configuration and use difficulties. 2. Does Asterisk 13 come with SQLite ready for use or have I to install this database separately and configure it to be accessible in dial plan? 3. Where can I find tutorials about using SQLite or the builtin database for storing simple that? P.S.: I'm not interested in storing CDR data. Any hint will be very helpful! Thanks a lot! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979(Brasil) -- _ -- 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
[asterisk-users] How may SIP 183 messages a caller receives when many callee rings?
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/2000SIP/2001SIP/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? Any hint will be very helpful! Thanks a lot. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 (Brasil) -- _ -- 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
[asterisk-users] RES: Banco de dados interno no Asterisk e variáveis em SIP HEADERS
Prezado Fernando,
Muito obrigado por sua complementação na resposta!
Surgiram algumas dúvidas agora:
A única forma de retornar os dados num header field, como o Rafael dos Santos
Saraiva sugeriu envolve criar outro channel?
Ou seja, o que eu preciso é que a mesma execução do dia plan obtenha um valor
recebido do Sip Client, execute uma query num banco de dados e em seguida
inclua a resposta como novo hearder field na mensagem a ser enviada de resposta
ao mesmo SIP Client.
Tudo isso pode ser executado no mesmo channel? Ou seja, sem precisar fazer um
Dial() para o Sip Client?
Por exemplo:
Suponha o seguinte, o SIP client envia um SIP INVITE para o Asterisk, contendo
um novo header field na mensagem. O dia plan executa, faz o que tem que fazer,
obtem um valor de um banco de dados e em seguida inclui esse valor como novo
header field na mensagem de resposta SIP ACK 100. Ou talvez na mensagem de
resposta SIP 180 (Ringing). Isso tudo seria feito num mesmo channel? O que
estou imaginando é usar as mensagem padrões SIP, que o Asterisk já sabe
manipular, e pegar 'carona' nelas para o transporte de pequenos dados.
Algo desse tipo é possível de ser feito?
No nosso projeto usaremos SIP com TCP, não com UDP, devido a outros requisitos.
Isso facilitará o uso da ideia com Json, certo?
Atenciosamente,
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
--
Só complementando a resposta do amigo Rodrigo.
O Comando SIPAddHeader vai adicionar um cabeçalho SIP, porém no channel
atual, e o Dial, criará outro channel, o qual não irá ter o cabeçalho
que você adicionou:
Se quiser que o cabeçalho SIP customizado esteja disponivel e seja
enviado para a Ponta B que o Dial está chamando, você terá que executar
uma Macro utilizando o canal novo que será criado pelo comando Dial.
Algo do Tipo:
[header]
exten = cid,1,SIPAddHeader(X-My-Header=MYCUSTOMHEADER)
same=n,Return(1)
[meudial]
exten = _X.,Dial(SIP/X.X.X.X/${EXTEN},,b(header^cid^1))
Porém, UDP tem suas limitações, e tentar incomporar JSON a SIP Message,
imagino que não consiga ter uma ambiente de fácil manutenção.
Uma ideia seria utilizar Kamailio ou OpenSIPs o que te da mais
ferramentas para gerenciar o SIP Message.
Ou você pode utilizar seu próprio esquema utilizando um sistema de
mensagens TCP como o ZeroMQ ou o GearmanD.
Atenciosamente / Best regards / Saludos,
P Antes de imprimir pense em sua responsabilidade e compromisso com o
Meio Ambiente!
-- Mensagem original --
De: Rafael dos Santos Saraiva rafaelsnsa em gmail.com
Para: asteriskbrasil em listas.asteriskbrasil.org
asteriskbrasil em listas.asteriskbrasil.org
Enviado(s): 12/06/2015 14:53:42
Assunto: Re: [AsteriskBrasil] RES: Banco de dados interno no Asterisk e
variáveis em SIP HEADERS
Rodrigo
Segue um exemplo de manipulação do SIP HEADER:
Servidor 1:
exten = _X.,1,Answer()
same = n,SIPAddHeader(Custom-variable: valor da minha variavel)
same = n,Dial(SIP/10.68.2.43/${EXTEN},30,tT)
same = n,HangUp
Servidor 2:
exten = _X.,1,Answer()
exten = _X.,n,NoOp(${SIP_HEADER(Custom-variable)})
exten = _X.,n,goto(ura,s,1)
exten = _X.,n,HangUp
Você enviar quaisquer valores que possam ser definidos numa variável.
Neste sites você encontra maiores informações:
http://www.voip-info.org/wiki/view/Asterisk+cmd+SipAddHeader
https://wiki.asterisk.org/wiki/display/AST/Home
O Jabber trabalha com o protocolo XMPP, de mensagens instantâneas.
--
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[asterisk-users] Can dial plan handle new proprietary SIP HEADER fields? How?
Dear asterisk-users, I have listened that a diaplan on Asterisk can extract information from proprietary SIP messages header fields. That is, if Asterisk receives a SIP message with a modified HEADER (containing proprietary fields) , is it possible to program the dial plan to make Asterisk extract the values of such fields, being possible to handle such values in diaplan, isn't it? If it is true, is it also possible to use dial plan to make Asterisk include proprietary SIP HEADER fields in a specific SIP message? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 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
[asterisk-users] Can Asterisk help me with some requeriments of my current project?
Hi Asterisk-user. I'm starting in a soft-phone project with lots of requirements and some of then caused me some doubts about Asterisk. Could someone tell me if Asterisk can help me with some requirements? See below: 1 - My SIP server (Asterisk) will have some SIP clients registered in its SIP registrar. Let's say 6 SIP clients. In my project I have to implement a way of a SIP client making a call to a number and all others 5 SIP clients ring. That is, the others 5 SIP clients must receive the SIP INVITE. Can Asterisk help me with such functionality? 2 - When several SIP client ring, if one answer the call first, the others will have to stop ringing immediately. Can Asterisk help me with this requirement? 3 - How to avoid one of the SIP clients receiving SIP INVITES? That is, one of the SIP clients is forbidden to receive calls. Is there a way to program it in Asterisk, maybe via dial plan? 4- Let's suppose that I have a data base (let's say SQLite) in my SIP server (Asterisk) and I need implement a way of SIP Clients executing queries in such database. Could such queries be done/sent via SIP messages to Asterisk? Is there a way of accessing a database by meas of Asterisk, during a call, for example to collect information about others SIP Clients? Here I'm intending to create a software to be a kind of interface between Asterisk and the database, if necessary. 5 - If I need to send SIP messages all encrypted, using SSL or TLS , to the Asterisk, will this SIP server be able to interpret all messages correctly? Is there a way of let Asterisk talk with SIP clients in a secure way, using SSL, for example? Can Asterisk help me with this? Any hint will be very helpful!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 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
[asterisk-users] RES: RES: RES: How to invoke a binary file from the dial plan?
Hi Kevin. Thank you again for help me! In my case, in the final application for smartphones or in a softphone for PCs, there will be a button on the GUI and the user will have just to touch it, and the door or gate will open. I mean, during an ongoing call, the callee will see a button in the interface of its SIP application. For example, we can use the lib of Linphone and implement a GUI over it, having a new button to open doors and gates. So, the callee will not have to remember about codes, because there will be a button in someplace to be touched. When the button be touched, during an ongoing call, the software (SIP client) will sends a request to Asterisk executes the gate = 9,self/callee,System,insert command here , for example. So, it will works like the user pressing number 9. I will take a look at applicationmap in features.conf to understand what exactly can be done. But, let me ask you: This idea seems to be good to run during ongoing calls. What about moments when there is no ongoing call? That is, can Asterisk execute a dial plan (maybe by means of some kind of SIP request received from the SIP client) even without establishing a call? Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] em Nome de Kevin Larsen [kevin.lar...@pioneerballoon.com] Enviado: quarta-feira, 3 de junho de 2015 10:29 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] RES: RES: How to invoke a binary file from the dial plan? Hi Kevin. Thank you very much for the hint! It worked very well! Your example ' exten = 1234,1,System(echo This is a test / var/log/asterisk/test.txt) ' executes when the SIP client (my softphone Jitsi) sends a SIP INVITE to asterisk. So, the softphone tries to establish a session with target 1234. Now, lets suppose my softphone rings and I answer a call. During the call, the caller asks me to execute a command (ex: to open a door or gate). In this case, what have I to program in dial plan to Asterisk execute System() again? Is it possible to execute a dial plan even during an ongoing call? Finally, lets suppose I want to use my softphone to execute a dial plan, even without establishing a call (no session with target 1234). For example, If I decide to open a dor or gate using my softphone, without existing an ongoing call, what have I to program in dial plan to Asterisk executes System(). Is this idea possible? Any hint will be very hepful! I love this question, simply because it allows me to talk about one of the neatest features I programmed into my system that barely anyone knows exists. Plus it lines up pretty much exactly with what you are trying to do. We have our gate control system tied into our Asterisk phone system so it is possible to dial a code on the phone and open the entrance gate to let someone in after hours. Only problem is this happens so rarely that no one (myself included) ever remembered the code. Thus a search for a better way. Now, when someone uses the gate phone to request entry, I change the caller ID on the display of the person who answers to read Press 9 to open gate. During the call, they can hit 9 at any time and the gate will open for them. Up until they answer, the caller ID reads Gate Phone, but when they answer, it changes to that text. The part about opening the gate is the magic piece you want to look into. Read up on applicationmap in features.conf. It's pretty simple and very effective. Here is what mine looks like. I am going to replace my actual command with insert command here. gate = 9,self/callee,System,insert command here ; Custom application to open the gate. This says that this feature is active in the 'gate' context of my dialplan. The dialing pattern it is looking for is a 9. 'self' tells it to activate on the channel that dialed it and callee says that the person receiving the call is the only one that can activate it (otherwise the person at the gate phone could hit 9 to open it). I am running the System dialplan application and passing it the insert command here value. Everything after the ';' is a comment as normal. The insert command here is equivalent to what you would put inside the '()' if it were in the dialplan (i.e. 'System(insert command here)'). Pretty straightforward to get it working once you know what to look for. Let me know if you want to know how I manipulate the Caller ID upon answering the call to give the instructions to the callee on how to open the gate/door. -- _ -- 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
[asterisk-users] RES: RES: RES: How to invoke a binary file from the dial plan?
Hi Kevin.
Thank you!
I will examine it.
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] em Nome de Kevin Larsen
[kevin.lar...@pioneerballoon.com]
Enviado: quarta-feira, 3 de junho de 2015 10:34
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] RES: RES: How to invoke a binary file from the
dial plan?
I love this question, simply because it allows me to talk about one
of the neatest features I programmed into my system that barely
anyone knows exists. Plus it lines up pretty much exactly with what
you are trying to do.
We have our gate control system tied into our Asterisk phone system
so it is possible to dial a code on the phone and open the entrance
gate to let someone in after hours. Only problem is this happens so
rarely that no one (myself included) ever remembered the code. Thus
a search for a better way.
Now, when someone uses the gate phone to request entry, I change the
caller ID on the display of the person who answers to read Press 9
to open gate. During the call, they can hit 9 at any time and the
gate will open for them. Up until they answer, the caller ID reads
Gate Phone, but when they answer, it changes to that text.
The part about opening the gate is the magic piece you want to look
into. Read up on applicationmap in features.conf. It's pretty simple
and very effective. Here is what mine looks like. I am going to
replace my actual command with insert command here.
gate = 9,self/callee,System,insert command here ; Custom
application to open the gate.
This says that this feature is active in the 'gate' context of my
dialplan. The dialing pattern it is looking for is a 9. 'self' tells
it to activate on the channel that dialed it and callee says that
the person receiving the call is the only one that can activate it
(otherwise the person at the gate phone could hit 9 to open it). I
am running the System dialplan application and passing it the
insert command here value. Everything after the ';' is a comment
as normal. The insert command here is equivalent to what you would
put inside the '()' if it were in the dialplan (i.e. 'System(insert
command here)').
Pretty straightforward to get it working once you know what to look
for. Let me know if you want to know how I manipulate the Caller ID
upon answering the call to give the instructions to the callee on
how to open the gate/door.
I just realized I said one piece wrong in this. 'gate' is not the context, it
is the dynamic feature designator. I can illustrate this better by posting my
front gate context.
[front_gate]
exten = number gate dials goes here,1,Set(__DYNAMIC_FEATURES=gate)
same = n,Goto(frontgate_queue,${EXTEN},1)
--
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[asterisk-users] RES: RES: How to invoke a binary file from the dial plan?
Hi Kevin. Thank you very much for the hint! It worked very well! Your example ' exten = 1234,1,System(echo This is a test /var/log/asterisk/test.txt) ' executes when the SIP client (my softphone Jitsi) sends a SIP INVITE to asterisk. So, the softphone tries to establish a session with target 1234. Now, lets suppose my softphone rings and I answer a call. During the call, the caller asks me to execute a command (ex: to open a door or gate). In this case, what have I to program in dial plan to Asterisk execute System() again? Is it possible to execute a dial plan even during an ongoing call? Finally, lets suppose I want to use my softphone to execute a dial plan, even without establishing a call (no session with target 1234). For example, If I decide to open a dor or gate using my softphone, without existing an ongoing call, what have I to program in dial plan to Asterisk executes System(). Is this idea possible? Any hint will be very hepful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] em Nome de Kevin Larsen [kevin.lar...@pioneerballoon.com] Enviado: terça-feira, 2 de junho de 2015 17:50 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] RES: How to invoke a binary file from the dial plan? Ok. Thanks for the hint. But, what exactly is a System() dialplan application? Is it a kind of command that i can call in dial plan? I will look for System() related to dial plans. From the Asterisk CLI type: core show application System It will print out the syntax for the command. One of the easier dialplan applications. exten = 1234,1,System(echo This is a test /var/log/asterisk/test.txt) That line would use the Linux echo command to place the text This is a test into a file named test.txt located in the /var/log/asterisk directory. -- _ -- 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
[asterisk-users] RES: RES: RES: RES: How to invoke a binary file from the dial plan?
Ok Kevin. Thank you for the information. Now, I will try to build a prototype to see how everything works. If I have a new doubt, I will post it here. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] em Nome de Kevin Larsen [kevin.lar...@pioneerballoon.com] Enviado: quarta-feira, 3 de junho de 2015 12:26 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] RES: RES: RES: How to invoke a binary file from the dial plan? Hi Kevin. Thank you again for help me! In my case, in the final application for smartphones or in a softphone for PCs, there will be a button on the GUI and the user will have just to touch it, and the door or gate will open. I mean, during an ongoing call, the callee will see a button in the interface of its SIP application. For example, we can use the lib of Linphone and implement a GUI over it, having a new button to open doors and gates. So, the callee will not have to remember about codes, because there will be a button in someplace to be touched. When the button be touched, during an ongoing call, the software (SIP client) will sends a request to Asterisk executes the gate = 9,self/callee,System,insert command here , for example. So, it will works like the user pressing number 9. I will take a look at applicationmap in features.conf to understand what exactly can be done. But, let me ask you: This idea seems to be good to run during ongoing calls. What about moments when there is no ongoing call? That is, can Asterisk execute a dial plan (maybe by means of some kind of SIP request received from the SIP client) even without establishing a call? The way I would probably approach what you want to do is that the button action state would be dependent on if you are in a call or not. If you are in a call, it sends whatever DTMF digits you want to use for this feature. If you are not in a call, it could dial an extension whose purpose is to do the same thing. I have an outside number that when dialed checks that your caller id number is in an approved list and if it is, sends the gate open signal. This is the same gate open signal that the feature code uses (the call to System()), it is just reached by making a sip call. Nothing says a call has to connect two phones together. You can answer the call inside of Asterisk and do stuff based on what number you called or what digits the caller enters with their keypads. Lot's of opportunity to make the system do exactly what you want. -- _ -- 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
[asterisk-users] How to invoke a binary file from the dial plan?
Hi everyone. I'm new with Asterisk and I have to create a dial plan that will invoke a binary code. That is, asterisk will execute a program in the same machine. How to do it? Let me explain what I have to do: In the project that I am currently working, there is smartphones, SIP servers and doors/gates to be unlocked remotely. When the user executes an application on his/her phone, it will presents a button to unlock a remote gate or door. By pressing such button, the application will send a SIP INVITE to the SIP server (Asterisk). In this moment, a existing dial plan should call an executable hosted in the current machine. In this case I need to know how to program my extensions.conf to let Asterisk invoke another software to me. The another software is the one responsible for unlocking a gate or door. So, how to codify my extensions.conf in order to make Asterisk invoke another software? Is another better way (idea) to implement my project using Asterisk and SIP? If so, comment, please! Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 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
[asterisk-users] RES: How to invoke a binary file from the dial plan?
Ok. Thanks for the hint. But, what exactly is a System() dialplan application? Is it a kind of command that i can call in dial plan? I will look for System() related to dial plans. Thanks. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] em Nome de Kevin Larsen [kevin.lar...@pioneerballoon.com] Enviado: terça-feira, 2 de junho de 2015 17:31 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] How to invoke a binary file from the dial plan? Hi everyone. I'm new with Asterisk and I have to create a dial plan that will invoke a binary code. That is, asterisk will execute a program in the same machine. How to do it? Let me explain what I have to do: In the project that I am currently working, there is smartphones, SIP servers and doors/gates to be unlocked remotely. When the user executes an application on his/her phone, it will presents a button to unlock a remote gate or door. By pressing such button, the application will send a SIP INVITE to the SIP server (Asterisk). In this moment, a existing dial plan should call an executable hosted in the current machine. In this case I need to know how to program my extensions.conf to let Asterisk invoke another software to me. The another software is the one responsible for unlocking a gate or door. So, how to codify my extensions.conf in order to make Asterisk invoke another software? Is another better way (idea) to implement my project using Asterisk and SIP? If so, comment, please! Any hint will be very helpful! Look into the System() dialplan application. It will execute a command on the system for you. Be aware that it will execute it as the user your Asterisk instance is running as, so permissions can sometimes be a bit finicky to get correct. I do something similar to pop my gate open. It is using nc to make a connection to the device, but same general idea as what you are doing. -- _ -- 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
