similar to: dial and bridge

Hi all, I have been playing with the AMI quite a bit lately - mostly debugging WombatDialer in production, but that's a different story - and I have been frustrated by the lack of a simple way to interact CLI-like with the AMI itself. So I have decided to write something myself to make my life easier, or at least a bit less miserable. The result is a little webapp that you can use as a sort

Hi all, I thought I'd share I script I made (based on some of Leif's works) that lets you download, compile and install Asterisk all in one go; and then removed the dev tools used. We use it quite a bit to provision systems using Ansible, but it is easier than remembering everything every time even if you are using a shell. At the moment I have scripts for Centos 7 and Asterisk 13, but

Hi there, I support a large number of enterprise users who contractually must connect to our support center via a 4G VOIP connection. I simply want to be able to auto detect all poor quality calls in realtme (as they are being made), play a message and drop the call - without user intervention. All decent call quality calls will be allowed through - to be handled by support staff. Its 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:

Hello all, I created a set of Docker images running Asterisk and exposing AMI / ARI ports that i found to be quite useful for ARI / AMI development and regression. As they are based on Docker with whaleware, adding new configuration files to roll your own dialplan / queues / voicemail etc is pretty easy. And you can run quite a lot on the same box to simulate clusters. There is no SIP / RTP

Hello Everyone, We are looking for a simple open source auto dialer with "polling" capabilities. What we would like is a program that we can upload leads to, and have asterisk: i) Dial numbers ii) Play pre-recorded iii) If user presses one, forward the call to an agent There are so many solutions out there it's hard to make a decision on what works, what has just a limited free

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hello list, I am trying to set a custom SIP header on all calls that are made by the app queue because I want to track a certain state at the SIP level. If I use the following code: exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID}) exten => s,n,Queue(myQueue) this works fine for the FIRST call made from the queue to an agent; but if that call does not go through, it's not repeated

Hey Guys, I knew this is stupid question but i just want to know what is the difference between Queue member logged out vs Pause ? -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110321/7646a66b/attachment.htm>

I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH

Hello all, I am trying to set up a dynamic channel to be used as an Agent dialer for a queue - you know, trying to replace AgentCallBackLogin for an Asterisk 1.6. I would like to do something like: [myagents] exten => XXX,1,Set(realchan=${DB(myagent/${EXTEN})}) exten => XXX,n,Dial(${realchan},tT,60) This basically fetches the actual channel to be used for dialling and dials it. What I

Hi Guys! We were using queuemetrics since long time with asterisk 1.2 but recently we have install 1.8 asterisk and but there is a big different in queue_log its saying SIP/XXXX instead of Agent/XXXX that is obvious behaviors. so do i need to change Agent/XXXX to SIP/XXXX in queuemetrics ? or is there any workaround to keep business running same like it was before. -S --------------

Hi all, I just prepared a little tutorial on installing Asterisk 13 on CentOS 6.5 64-bit. See http://astrecipes.net/index.php?n=668 Hope you like. :) l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.com

Hi, I am using RT engine to log queue_log to a mysql database. My extconfig is [settings] queue_log => mysql,asterisk16_production Logging to mysql is working fine. But I find that the queue_log file now only has QUEUESTART lines for eg: 1250519094|NONE|NONE|NONE|QUEUESTART| 1250519186|NONE|NONE|NONE|QUEUESTART| How can I have queue_log in both db as well as in a file? thanks and

Hello Everyone, I was wondering how SIGTRAN/SS7oIP can fit into our currently 100% SIP model. We are looking to interconnect with the PSTN world, and our supplier has given us a few options. We can either do this over traditional PRIs, A-Links or the SS7IP new. I am really interested in SIGTRAN, and was wondering how some of you have integrated it into your architecture. Can Asterisk handle

Hi everyone, after doing the same thing multiple times and struggling to remember how it was done, I have prepared a small tutorial that explains how to save monitored files in different folders per day. This is quite useful becausethe resultingfile system is way more manageable than having maybe 100,000 files all saved in the same folder. You can find the tutorial here:

Hi, If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC video from one machine to another machine running Linphone. Contact me at this e-mail address robkrakora at messagenetsystems.com for source code. Best Regards, -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032

Hello list, I have been working lately on a small CDR parsing utility, and would like to do some performance testing on it. I am looking for some - possibly large - real-life Asterisk CDR datasets to run some performance monitoring. Anybody's got some CDRs that can be shared? Thanks in advance, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com

Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) l. -- Home of QueueMetrics -

Hello List, In PHPAGI, I'm using the Astrisk Manager function send_request() to originate an outbound call. I want to execute the remaining PHP code after the call gets executed (depending on user input). But presently the call originates in a different context and asterisk executes the remaining code in parallel. Is there a way in which I can pause the code execution until the call is