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...caller. What do you think? Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 (Brasil) ________________________________________ De: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] em Nome de SamyGo [govoiper at 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 m...

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>

....wait5,n,Hangup exten => 555,1,Dial(LOCAL/6001&LOCAL/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 <govoiper at 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/...

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>

...tation. 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-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] em Nome de SamyGo [govoiper at 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 u...

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

...lude a new queue or agent I need to restart the app_queue, than the asterisk lost this informations and begin with an empty set. there is a way to resolve this? if anybody knows please give me this informations or hints to revolse this... thank's a lot for the opportunity Everton Goularth GOVoIP www.govoip.com.br _______________________________________________________ Yahoo! Mail - Sempre a melhor op??o para voc?! Experimente j? e veja as novidades. http://br.yahoo.com/mailbeta/tudonovo/

...> cdr_odbc: Connected to oracle > cdr_odbc: Trying Query again! > cdr_odbc: Error in Query -2 > cdr_odbc: Query FAILED Call not logged! Does anyone have any ideia? Did anyone have this error?? or know how can I do this? Thank's in advanced... Everton Goularth GoVoIP - Uberlandia - MG Brasil _______________________________________________________ Yahoo! Mail - Sempre a melhor op??o para voc?! Experimente j? e veja as novidades. http://br.yahoo.com/mailbeta/tudonovo/

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 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 =>

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 ?

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.......

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. 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 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",

Hi everyone, Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about 5000 numbers and then put the call to agents right away and pull up the CRM based on the number dialed. So, I am going to be doing some PHP+Ajax work. I am familiar with spool files but I don't like the fact that I can't read the status of the call in real-time. However, I know that it's the

Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing .... when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not

HI All asterisk user I Have one Asterisk with this scenario: i have two ip Address one Private IP one Public IP, my internals terminals using private IP works very fine but my terminals using public ip don't work audio , make rings but streamer don't work. thks for you attetion best Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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