similar to: Messages out of calls. Is it really possible?

Hello My provider allows to activate/deactivate a forwarding rule by sending a SIP MESSAGE. This is done outside a call. That is, while there is no ongoing call, a SIP client just sends the following message: MESSAGE sip:543951354657 at callfwd.sip.providerx.com SIP/2.0 Call-ID: b9ba106e-613a-46b9-8a4d-0efb4dc0a0f2 CSeq: 1 MESSAGE To: <sip:543951354657 at

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

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

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. 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 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 =

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. 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,

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)

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 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

On 15-09-28 10:19 AM, Emil Ohlsson wrote: > (Still no not receiving the mail, revisited the settings.) > > OK, so SendText doesn't work with this scenario. But can MessageSend > handle this, and respond even when the transport protocol is TLS? Or > do I need to modify Asterisk to add this support? MessageSend has no concept of TLS, it gets passed to chan_sip which then sends

On Mon, Sep 21, 2015 at 9:45 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote: > On Mon, 21 Sep 2015 06:48:52 +0000 > Emil Ohlsson <emo at svep.se> wrote: >> [sip-im] >> exten _X!, 1, NoOp(Got message) >> exten _X!, n, Answer() >> exten _X!, n, Agi(agi://localhost/messagehandler.agi?...) >> exten _X!, n, SendText(Message received) > > I am not

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

Sorry for the delay here. For some reason the mail from Joshua Colp failed to deliver to my mailbox. So, anyway, I've set up a local scenario on my computer a PJSIP client and Asterisk 11.17.1 (On a fedora linux workstation) with the settings listed below. In this scenario I've used UDP, but I want a configuration that can be used with any transport protocol. I can see that the context

Hi, I'm having trouble configuring Asterisk to respond to an incoming out of call SIP MESSAGE. The transport protocol is TLS and the Asterisk version is 10 (it's old, but I'm kind of stuck with it at the moment). Currently I have roughly the following configuration and handling: sip.conf: [general] accet_outofcall_messages=yes outofcall_message_context=sip-im and extensions.conf