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I am running into an issue with A2Billing. I will explain first of all that everything else works! the system is 90% complete its just this one small problem I am running into. So my problem is that when I place a call, 1. I dial my number that I want and A2Billing gets activated 2. it asks for my pin, upon successful entry of my pin A2Billing then 3. prompts me for my phone number then 4.

Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after

Hello, I am working on a trixbox based system with a TDM410P connected to 3 phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN with some polycom and Aastra SIP phones. In general everything works. the problem I am trying to solve is that if both parties to a call speak at the same time one of the voices gets cut out such that the talker A cannot hear what talker B is

Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or Local/1234 at ext) and "park" it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the

After working professionally with Ruby on Rails for a year, I decided to write an article on my experiences with the framework. http://nathany.com/developer/rails-magic Since I detail a number of things that I found unintuitive or could be improved upon, I am posting a link here on the Rails Core in hopes to stimulate David Heinemeier Hansson and the core team towards an even better 3.0

READ READ READ .... http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Sun, Feb 22, 2015 at 8:25 AM, thufir <hawat.thufir at gmail.com> wrote: > I'm looking into the dialplan specifics:...

With that kind of load, your users shall start complaining about choppy audio or voice clarity on random occasions, and you wont have a clue where to look for the problem. Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Tue, Apr 7, 2015 at 9:57 PM, Jeff LaCoursiere <jeff at jeff.net> wrote: > On 04/07/2015 10:48 AM, Johan Wilfer wrote: &...

Hello Doug, Maybe you can have it uploaded on GitHub.com as a repository ? With a README.md file on how to install it for PHP7 ? Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Mon, May 25, 2020 at 3:17 PM Doug Lytle <support at drdos.info> wrote: > Everybody, > > I've been using the...

Hi, I am having some trouble setting up Caller id on my asterisk system, I need to know if there is anything special that needs to be done for an australian connection specifically as I have tried what most web sites on google reccomend but without success. I have not had much experience with asterisk as I have inherited this system from the previous sysadmin who has not documented anything so I

...really started all the fuss with Ruby and Rails, Dave Thomas. We also have a bunch of quality speakers including Jim Weirich, Brian Sam-Bodden, Joe O''Brien, Jeremy McAnally, Gregg Pollack, Glenn Vanderburg, Neal Ford, Andrea O.K.Wright, Jeff Barr, Kyle Banker, Corey Donohoe, Nick Sieger, Nathanial Talbott, Jon Larkowski, Aman Gupta, David Bock, and Joe Damato. And we have lots of great local speakers, like Paul Barry, Dave Bock, Jeff Casimir, and Russ Olsen. The topics are great, too. They stretch from getting under the covers of the MRI, to NoSQL solutions, to web services with Ruby, to...

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

Hello, I have a fully functional Asterisk Server, I want to configure this server to be able to process call from Skype, can someone point me to a howto? or if there are suggestions on best way to approach this problem. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have been using Red-fone foneBridge2 Quad T1/E1 for last few years. As these devices are not available anymore, we are looking for alternatives. Are there any similar devices available ? -- Regards, Tirveni Yadav www.udyansh.org What is this Universe ? From what it arises ? Into what does it go? In freedom it arises, In freedom it rests and into freedom it melts away. Upanishads.

Hi there, I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine :-) But I am wondering if there is a solution/application which will enable me to implement voice recognition while playing a voice file (barge in). So that the caller hears a voice file and can interrupt it with his voice. Currently (on our platform) the caller has to wait for the end of the voicefie. Then we play

Am looking for a service provider who can provide enterprise SIP trunk with 100 channels concurrent sessions. I see some like Inphonex, Broadvoice... and etc.... Is there any suggestions for the service providers. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Everybody, I've been using the old Asterisk CDR Areski GUI CDR-Stats for at least a dozen years, it was easy to configure and didn't requite installing 'connectors' on anything or adding tables on the DB server. It's based off of PHP5 and the only reason I still keep around a Debian 7 system, since it won't work with the newer PHP7. A friend of mine is learning PHP7

Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device

Freezing to the gems for Rails 1.2.6 rake aborted! uninitialized constant Gem::GemRunner tracing gives: no such file to load -- /Users/nathany/Sites/boat/oat/config/../vendor/ rails/railties/lib/initializer /Library/Ruby/Site/1.8/rubygems/custom_require.rb:27:in `gem_original_require'' ... which seems like it may be trying to load the frozen gems before they are there? It

If SIP channel driver needs to connect to a remote GW over a dedicated SIP trunk BUT the remote GW has a 'standby' in case of failure, how can the sip configuration file be configured for the remote GW when there are actually two IP addresses. If the main remote GW fails control automatically switches to the standby GW, so how could the SIP configuration file hande this switch and support