similar to: How to get presence using AMI

Dear list, I am looking for an open source SIP client(or any SDK) that can work on a browser. It may be based html5, javascript, flash, adobe air. I have done some research myself and I would like to ask the community if they have any further hints for me. Real life experience would be awesome. Thanks, Regards, Arstan Jusupov -------------- next part -------------- An HTML attachment was

Hi, I'm writing a node.js backend to pass events via a websocket to a CRM. Basically what I want to do is notice when things happen (i.e. new channel, new bridge etc) without sending the channels to the Stasis app. The channels I'm interested in are agents who are in a queue only because they are in a realtime MySQL database for the queue_member_table. There doesn't appear to be a

>>I'm not sure that your question is clear. You'll probably want to be more specific. >> What is pulse? You mention "as a user", are you talking about voicepulse.com ? >> What are you trying to do with pulse? >> What problem are you running into? Sorry Rusty... I am trying to get Asterisk 11 to co-exist with a centos 7 box that has pulse audio running as

Ah, no, you misunderstand. Asterisk wouldn't care one little bit what is on the page - Chromevox would do all that. A screenreader usually tabs or arrows their way about, selecting headings to read content. Thus, Asterisk ONLY needs to be able to hear content FROM the browser and pipe it to the channel, and pass keypresses back TO the browser. The human is the parser, if that makes sense?

?I meant how many calls are in progress on a particular trunk. (Sorry - I didn't even think of the other interpretation). ________________________________ From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Matt Riddell <lists at venturevoip.com> Sent: Sunday, June 28, 2015 9:26 AM To: Asterisk Users List Subject: Re:

Hello, I'm using an AMI Originate command to send a fax. The fax is sent by a script, and I'd like my script to send the fax, wait until it has succeeded or failed, then exit with an appropriate error code (it is driven by a mail system, so the exit code will tell the mail system whether to retry the fax later). The script works great if the fax succeeds, or if the line is busy or

Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? I suspect we will have to build an AGI script but I'm hoping something new in Asterisk 13 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150627/6774c750/attachment.html>

> On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz at gmail.com> wrote: > > forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. > > Thanks, > > > On 04/27/2015 02:38 PM, Motty Cruz wrote: >> here is what I have: >> exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) >> >> exten =>

Hi Does anyone used asterisk as a operator phone,with multiple lines and features like transfer forward and etc.I used chan_alsa driver to make asterisk as SIP Phone,but it has limitation,we cant make or receive multiple calls,and will not able to do any features like transfer forward etc. Is any other application available in asterisk to do this . Thanks Nikhil

I am originating a command via the AMI with this... Action: Login Username: xxx Secret: yyy ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: Local/6505551212 at LegA Callerid: 849120 Context: default ActionID: 849120 My LegA context: ----------------------- context LegA { _X. => { Dial(SIP/${EXTEN}@Provider); } } And my default context:

Hello, I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with success. I have an application that sends an action Originate to AMI for calling, it's working well, but when i see to Asterisk's CLI, i see 2 calls for just one originate: pftestes40copiabh*CLI> core show channels verbose Channel Context Extension Prio State Application

Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil

Hi everyone Anyone please explain me How Account code is use for billing., Thanks Nikhil

hi, you still interesting in it? that I made long time ago. http://lists.digium.com/pipermail/asterisk-dev/2010-March/043096.html also I keep another patches and things and I need dedicated ftp for it. if you can give me such things I'll provide this patch to you. On 3 February 2011 09:44, Nikhil <d.nikhil at cem-solutions.net> wrote: > Hi everyone > Any one used Radius based

Hello, I am using Asterisk 1.4 version. How to dial multiple numbers per second through asterisk manager???? Thanks and regards

Is it well known that setting the ActionID when connecting to AMI has absolutely no effect? Is this fixed in Asterisk 1.4? If you add an ActionID to your Originate command for example, it looks like the only events that come back with an ActionID associated are the initial response, OriginateSuccess and OriginateFailure. That's it. No other events have an ActionID associated. This pretty much

Hello All, I'am a new Asterisk user, and i have the following question. The following is the Status of all open channels from my Asterisk system, which was received through the Asterisk Manager Interface ((AMI)). ==================================================================== action: Status actionid: 65066874_3# Response: Success ActionID: 65066874_3# Message: Channel status will

Hello, I am trying to stream an IP Camera output (h264) into a conference. The IP Camera supports RTSP. Searching around the web, I believe the RTSP support (was) available through app_rtsp (external to Asterisk distribution). This, I believe, has problems and has issues compiling in Asterisk 11 (I tried compiling it in Asterisk 11 and it failed). I may not be able to use DiaStar or i6net's

Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and

Hello, (sorry for not continuing the thread, I had set the list to digest). Would UnicastRTP be able to output u-law frames directly? If so, I think that is all I need. Does anyone know what the EAGI output is? Raw RTP? Best regards, Luca -------------- next part -------------- An HTML attachment was scrubbed... URL: