similar to: 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/6001&LOCAL/6002.wait5)

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

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

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

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

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

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

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

Hi. I have a beginner conceptual question about Asterisk: Let's suppose that there are 4 softphones registered in my Asterisk and all of them are currently online. In addiction , there is no call. Suddenly, one of these softphones sends a SIP message to the Asterisk. In this case the dialplan will execute the instruction ' exten => 2005,1,Dial(SIP/2000&SIP/2001&SIP/2002,

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

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-bounces at lists.digium.com

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

Hello Mynum: 6001 , Othernum: 6002. I can record as follows. But I do not enter individual records for each internal required. I want to do it more smoothly with a Macro. Thanks. exten => _6001,1,NoOp() exten => _6001,n,MixMonitor(${UNIQUEID}.wav,ab) exten => _6001,n,Dial(SIP/6001,20) exten => _6001,n,StopMixMonitor() exten => _6001,n,Hangup() On Sat, Mar 9, 2019 at 6:50 PM

Hi, Is there any way of telling Asteirsk to withhold answer subversion on a call till I call Answer. My DP looks like this: [incoming] Exten => 18005551212,1,Noop() same => n,Answer same => n,Mset(__uid=${SIPCALLID}) same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV) same => n,Dial(Local/1 at dial_call_center/n&Local/2 at dial_call_center /n&Local/3 at

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

> 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

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

Hi all, I'm required to send a dialplan variable with every AMI event triggered for the duration of the call. For example; ... exten => s,n,Set(MyVar=${ODBC_GetSomething(${EXTEN})) ... so can I have this Variable MyVar attached in all AMI events for this call ? I can understand that untill this variable has not been set some value it may even be empty but as soon as its set I expect some

Hi, I've successfully setup Asterisk on my local PC and can make call using Twinkle to the server. But, I cannot call to my Asterisk server at Rackspace. I have been trying several things to figure it out, no luck. My PC is behind NAT, so I've set that up in sip.conf (nat=yes). I can ping my Rackspace server so it seems to be Public-static IP. Anyway, I tried with setting externip,

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems