similar to: Adding Variable in all AMI events

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

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 All, I've been wondering if I can instruct asterisk in the dialplan to engage the Media handling for a particular call or not. I've SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf setting "directmediadeny|directmediapermit" to offload media from asterisk for peer-to-peer calls BUT what if someone wants to record a call or engage some feature-code ?

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 trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

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,

Hi, I write some php code in AMI to working with asterisk command. I don't know exactly what is the different between AMI and AGI and witch one is better for my planning. Im planning to call party users that their number is is my panel on web. We have some operator and they can call party users via client softphone by clicking on their number, so they have to limited to call just listed

Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I

Hello, Is there a way to either store login/logout agent information in a database or at least send an email when an agent logs in or out of a queue? Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110812/84130e1a/attachment.htm>

Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111214/b462516a/attachment.htm>

Hi list, What possibilities are there in asterisk to send an *alphanumeric DTMF*from/to asterisk !? Regards, Sammy -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120228/e62e7890/attachment.htm>

Thank you very much, Max, for this valuable and informative answer. Offline billing must be quite complex to set up as several telco may be involved (or origination,transit or termination). Moving to normal landline fare seems much simpler ! Thanks again 2017-03-28 21:41 GMT+02:00 Max Grobecker <max.grobecker at ml.grobecker.info>: > Hi, > > in Germany, this kind of regulation

Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: ---------- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the

Hi team, I am trying to find solution to hangup b-party call after 1 min with out disconnecting the call of a-party but following dial plan which is disconnect both the calls. Please suggest me the solution. [TB] exten => _X.,1,Wait(${INCOMING_WAIT}) exten =>_X.,2,Verbose(TB) exten =>_X.,3,Answer() exten => _X.,4,Set(mainLoop=0) exten =>

Hello, I'm using AGI scripting with asterisk and need to execute certain commands just after Dial(). But once dial command is executed, further commands/instructions are ignored. $agi->exec("Dial","SIP/100"); $dialstatus = $agi -> get_variable("DIALSTATUS"); if($dialstatus[data]=="ANSWER") { do something.......

Dears; In asterisk 1.8, it is not more possible to use DeadAGI? Also, I found the below commands in the a2billing and I would to ask why it set the sequence 1 for the Hangup()? Maybe because it is related to the NoOp? How? [a2billing-callingcard] exten => _X.,1,NoOp(A2Billing Start) exten => _X.,n,Answer() exten => _X.,n,Wait(2) exten => _X.,n,DeadAgi(a2billing.php,1) exten =>

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before

Hi list, I want ot do basic work (add-edit-delete) into asterisk configuration files, like sip.conf, manager.conf,musiconhold.conf etc. Please guide me how to configure all these files from from AMI connection. I am able to login into AMI from Login action but I want to do more task in to it. *AMI login:- * *login.php* <?php $socket = fsockopen("127.0.0.1","5038",

I'm trying to create a new column in my data.frame where subjects are categorized depending on values on four other columns. In any other case I would just nest a few ifelse statements, however, in this case i have 4*6*2*3=144 combinations and i get weird 'context overflow' errors. So I wonder if there is a more efficient way of doing this. For illustrational purposes, let's say