similar to: html/js/flash/air SIP clients?

I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten

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:

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 =>

Topic: Phono and Phono SDK, an exciting development you can read about at http://phono.com You can learn all about the technical aspects of Phono and the SDK Friday with Chris Matthieu, but here are a couple of interesting implementations that don't require much effort to show a proof-of-concept: 1) There is already a WordPress plugin that literally allows you to add a button on your

Greetings, I want to place a form on my site so customers can recieve an mmediate callback and the PBX should connect them to a cell sales agent. Are there anfree modules available for this, or one should code this from scratch? What I want is when a potential client submits his number... the PBX dials the number makes an announcement and dials an extension (which is actually a cellhopne dahdi

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

Hi Using AMI how can I get the presence feature.Below are the requirement. --> List of all users in the PBX including analog and SIP including registration status. --> Status(BUSY or available ) of all users both analog and SIP Please help on this.. Thanks Nikhil

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:

On Thu, Jun 18, 2015 at 1:26 PM, Matt Riddell <lists at venturevoip.com> wrote: > Did you buy the number from your carrier? Maybe it?s set on their side > for the trunk. > That's what I think too, but they are denying this. I think what's happening is they have a customer service guy interpreting logs (probably incorrectly). When I had a Century Link POTS line, I had a

Hi All, I wonder if any of you has some suggestions on which WebRTC client/softphone to use for a click-to-dial, webpage hosted solution. Any suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Wed, 31 May 2017, Daniel Tryba wrote: > On Wed, May 31, 2017 at 01:39:25PM -0700, Steve Edwards wrote: >>> What bugs you about the output format? >> >> It's been a while, but as I recollect, it included the date/timestamp in the >> file name of the 'ring buffer' which meant that each time the host was >> rebooted, dumpcap didn't know the

Hello, I have been working on designs for two different projects, where both of them would need to use the IBM Watson streaming ASR service. Based on our discussion at AstriDevCon, I know there is currently no support for that. However, there may be some workarounds I am not aware of. Would it be possible to write out the audio frames as they get recorded? Watson supports 16 bit signed little

Hello I have already several Asterisk servers running with similar configuration, but now I stumble into a problem. I have mysql configuration res_config_mysql.conf : [MyAsteriskDB] dbhost = 127.0.0.1 dbname = MyAsteriskDB dbuser = astadmin dbpass = mysecret dbport = 3306 dbsock = /var/lib/mysql/mysql.sock requirements=warn ; or createclose or createchar Realtime seems to be loaded :

Thanks very much for all the responses. I now have a few more things to try. I should have noted that I am using IAX2 rather than SIP to connect to my provider. I do have some internal phones that use SIP to connect to my asterisk box, as well as some corded phones connected through a Digium DAHDI-driven card. I am certain that the old number that is showing up as my caller ID is not present in

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