similar to: AGI - Ways to create a call

Hey guys, I'm having a really huge problem, it seems like Asterisk is locking my licenses of g729 after being used. For example, 10 people make calls using this codec. Then I can see the channels and the codecs being used, cool. But then when they hang up the call the codes are still there, as being used... I don't know if its a bug or a miss configuration, but the fact is that I've

Hi guys!! This is something that have always bother me, hope you can help me... :) I've 8 server connected using IAX / DUNDi, it works just fine. However, sometimes when some of our links goes down the server takes forever to appear back as OK at DUNDi's list and people can't call the other Box. It's happening right now: CLI> dundi show peers 00:14:22:16:54:c5 200.X.X.6

Hi masters! I've this Asterisk 1.4.15 running. yesterday I had to change the firewall schema that I had before. I use to have a FW that would be my network FW/Proxy and do the NATs for Asterisk. This FW was receiving too many requests from my LAN and it was making the Asterisk 'cut' the calls or reach very high latency. Yesterday I added a new FW just for this Asterisk. The same

Hi all, I don't know if its the right place to ask, but... Does any one have the asterisk-stat-v2 running with PHP5? Tks! -- Tiago Durante ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., Perseverance is the hard work you do after you get tired of doing the hard work you already did. -- Newt Gingrich

Greetings listers, I have been running 1.4.21 for about 7 months now, but have been told I have to move up the 1.4 food chain or into the 1.6 chain because 1.4.21 is too flaky for our POTS line handling (does funny things with echo, doesn't connect to external conference calls, etc.). Which release will give me the most joy/least headache/closest performance to

LOL - You mean he actually 'met' Newt Gingrich? How dare you not extend him credit!!! I mean seriously... For such a distinguished individual... Hey, not only have I met the heads of several multi-billion dollar corps, I have gotten absolutely blasted drunk with them. So I should get credit, a 40% discount, and your daughters phone number, right??? LOL Seriously, though. I think it

Dear List wanna configure click2call in such a manner that my asterisk box call two mobile numbers and connect both numbers to talk. I have configured voip gateway, my internal and external calls are working fine. please help , abhi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm trying to send an email with the voicemail details and I want to send a HTML link on it to make a click2call to the voicemail main, but the email is send with 'text/plain' encoding and thus it will not show the link, but the HTML in plain text on the body of the email, How can I change the enconding to 'text/html' so the link will get displayed correctly?

Hi, I have a PHP code with AMI to using in click2call system. here is my code: $user = "usernamr"; $secret = "secret"; $channel = 'SIP/' . $sip; $context = "from-internal"; $waitTime = "20"; $timeout = 20000; $priority = "1"; $maxRetry = "2"; $pos = strpos($number,

To the Centos maintainers: We tried to install Centos 4.4 i386 (i686 / Athlon) on a Dual Opteron 254 Iwill DK8N MB, 1GB (4 x 256M) system with a 3ware 9500S-4 and 4 x 80 GB disks in a RAID 10 array. Both tries caused Centos to first kernel panic until we turned off IO-APIC in the BIOS. Both tries also only loaded the UP kernel and did not load the smp kernel on the machine. The smp kernel was

Olivier, I think you are getting confused. Call me on 212-203-4357 and I'll answer your questions but basically I think you are doing this the wrong way. Regards, Dean Collins Cognation Pty Ltd dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ________________________________ From: asterisk-users-bounces at

Hi there, Just to announce that I've improved upon a greasemonkey script which allows users to dial any number (in the given regex format) by turning it into a clickable hyperlink. The script uses greasemonkey's ajax callback to a simple php controller script, so that the click does not navigate away from the current page. It requires an Asterisk Manager connection. See

- ActivaTSP can't work with Astmanproxy as Asmanproxy needs to be patched, - Asttapi wouldn't terminate a completed call. Which option would you pick ? Is there any other option (free or commercial) for Outlook click2call ? Best regards ------=_Part_283_12644120.1196417210166 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline

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Hi All. I'm running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS machine (2.6.32-358.18.1.el6.i686). As a client I'm using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m. I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call

I would add the below: -Recommend using CentOS 4.0 -Use squid rpm, no tar (this is for new users I'm guessing). -Recommend using etherape and iptraf (available as rpms) for a graphical overview of traffic. http://etherape.sourceforge.net/ -Recommend the use of chkrootkit, and TCP Wrappers (at the least put ALL: ALL EXCEPT PARANOID in /etc/hosts.allow) to protect servers. -Provide some

Full_Name: Allan Stokes Version: 2.8.1 OS: XP Submission from: (NULL) (24.108.0.245) I've just spent a hellish six hours trying to create my own R package with a bare bones "hello world" R function inside. I was able to create a package.tar.gz file eventually with much perseverance. My remaining problem is that when I try to install my simple package under RGui, it hangs.

Hello I'm trying for several days now to get ICE support for my Asterisk 11.23 on CentOS 6. My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230 --> softphone Zoiper (problem : no audio) Reverse does not work either. (problem : failed get local SDP) I followed this guide : https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

Hi, I'm evaluating possible design for a specific CTI application. CTI client is a fat client (it's a customer's requirement) which exchange data with a CRM server (build on mainframe). CTI client must : - display custom view mixing ongoing calls, presence and some user preferences (such as this user has forwarded his calls to his voicemail) - request call origination (click2call

Folks, I am curious to know what the best way to build click2call with asterisk? There are a bunch of examples of the web that use socket to launch first leg of the call and then dump the call to a context that dials the second leg of the call. Unfortunately, none of the solutions I found explained how to get the call status of the first leg. What if there is some issue with the channel, what if