[asterisk-users] FOP2 in Digium repository?
Hello everyone, I see that the yum install freepbx from Digium repository actually installs the latest FreePBX which is nice. However, I don't see the old FOP in FreePBX anymore. Is there a way to install FOP or FOP2 through repository? Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing script
We have had success with this: https://sites.google.com/site/a2billing2asterisk/ On Fri, Mar 16, 2012 at 1:28 PM, Tahar .H harazta...@gmail.com wrote: hi folks, i was wondering if some one has a2billing script,which can be used to install a2billing easly ? thanks in advance -- * HARAZ Tahar * *Engineering Student at the National Institute for Posts and Telecommunications (INPT) * * Phone: +212 6 78030050 E-mail: harazta...@gmail.com ouabimedcha...@gmail.com * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rate sheet normalization
I would be more interested in a system where quality routes are tested with different providers because rate really doesn't matter if a call can't be placed or if a destination is a fake one. We have seen many fake destinations with top tier providers but they had the best rates so the strategy to pick them first really didn't work. So, maybe a subscription service where a dialler system continuously tests routes with a list of 10 providers so that it's established which routes actually work and then allow that data to be downloaded for usage. On Thu, Mar 15, 2012 at 8:42 PM, Markus unive...@truemetal.org wrote: Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर): On Thursday 15 Mar 2012, Markus wrote: With like 10 different ratesheets from 10 different providers, of which many change their rates every few days, manually doing it in Excel is too time consuming... Is it possible to get samples? I'd be interested in looking into developing a script that can handle this problem generically, and presumably you're available to alpha- and beta-test in any case :) Most definitely! I'll get in touch off-list. :) -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rate sheet normalization
A2Billing doesn't do that. A2Billing in fact has a lot of shortcomings one of which is this exact issue. I would suggest running rate sheets against each other for finding true LCR and then only uploading the rates that are cheaper into the system. In most cases there are not such high differences but if there are then this is the only way. I know rate normalization talk comes up all the time on FreeSwitch Freenode channel and it probably does on OpenSIPs as well. Check there for some good advice. On Wed, Mar 14, 2012 at 7:35 PM, Benny Amorsen benny+use...@amorsen.dkwrote: Markus unive...@truemetal.org writes: Does such a thing exist? How does a2billing do it? It should be pretty easy in an AGI. If you can afford a linear lookup per call, just grep through the array of prefixes to find the ones matching a particular call, then pick the cheapest from the results. If you need something faster than linear it gets tricky. It would be tempting to preprocess the list to say 5 digits, do a hash lookup on those, and then use the process above. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.9.3 Problem With Logger
Our experience has always been, with all versions of Asterisk, that when you do an originate command from Asterisk CLI then it stops showing CLI verbose and one has to open a new terminal to see the verbose. Logging status is in disarray in all version. These behaviours breaks security tools depending on logs. Best, On Thu, Mar 8, 2012 at 4:41 AM, Jon Farmer j...@bctech.co.uk wrote: Hi Just realised this is due to a FIFO blocking. Fixed that and all back to normal. Regards Jon Jon Farmer Tel 07795 118140 On 7 March 2012 16:33, Paul Belanger pabelan...@digium.com wrote: On 12-03-07 04:29 AM, Jon Farmer wrote: Hi I have recently upgraded a box to 1.8.9.3 and have noticed that randomly the logger will just stop working. It stops logging to the console and to the log files. Reloading logger actually freezes the console. It seems to happen when Asterisk tries to rotate it's log files but it may happen at other times too, The only way I have managed to get it back is to kill and restart asterisk. Any ideas what is going on. I'm pretty sure there is an existing issue in JIRA about this. Try searching it first, if not open a new issue so we can triage it. It is likely a deadlock so attach the required information for it. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up
Downgrade actually worked fine in this instance - from 1.8.9.2 to 1.8.9.1 and I concluded it wasn't Asterisk that was the issue. Thanks Patrick. It would be great to keep this feature stable. As it helps in case of regressions, etc... On Mon, Feb 27, 2012 at 10:09 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/27/2012 09:05 AM, Jason Parker wrote: On 02/26/2012 06:22 PM, Patrick Lists wrote: On 25-02-12 19:47, Jason Parker wrote: yum and rpm do not support downgrades. Incorrect. There is yum downgrade. See man yum. yum downgrade is extremely broken. It fails, often, potentially leaving a system in an unrecoverable state. That is not to mention how poorly conceived the concept is. Consider what would happen if a package upgraded some resource to a non-backwards-compatible version. It is completely unsupported on the Digium repositories. Please don't try it - I will not help fix it. This is generally true of *all* software packages; the developers/maintainers provide mechanisms for upgrading, but not downgrading. simple downgrades should be possible, but the packages aren't marked in any way to indicate whether that could even be done. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up
Thanks Jason. One more question: Is there anyway to go back on an Asterisk version when using the repository? For example, Asterisk 1.8.9.2 is available now. But I want to use 1.8.9.1. Can I downgrade somehow? I want to test NAT bug issue. Thanks On Thu, Feb 23, 2012 at 11:15 AM, Jason Parker jpar...@digium.com wrote: On 02/23/2012 10:09 AM, Ast Coder wrote: Hi, I have followed instruction on https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-Prerequisitesto add Digium Asterisk repositories but doing a, yum search asterisk only shows me Asterisk 1.4, 1.6, and 1.8. There is no Asterisk 10 and yum install asterisk10 fails. Am I missing something? or Asterisk 10 is just no available in binary? Thanks, There are now repositories for each major version of Asterisk, which have to be explicitly enabled to use them. `yum update` to get to the latest of everything, then do `yum update --enablerepo=asterisk-10`. Asterisk 10 will be installed, and that repository will be enabled permanently. I'll add that information to the wiki shortly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up
Hi, I have followed instruction on https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-Prerequisites to add Digium Asterisk repositories but doing a, yum search asterisk only shows me Asterisk 1.4, 1.6, and 1.8. There is no Asterisk 10 and yum install asterisk10 fails. Am I missing something? or Asterisk 10 is just no available in binary? Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Where can I find some good examples of listening to AMI events via PHP how to listen to a specific event?
Hi everyone, I got HTTP AMI working fine here. For example this dials 1-415-999- and then sends to Extension @from-internal: http://192.168.0.100:8088/asterisk/manager?action=commandoriginateDAHDI/g0/1415999extension@from-internal However, I want to have some control over this call. I want to be notified the moment this call is hangup. I guess there would be a hangup event generated. I am not sure if that would be done through action:waitevent? or if there is another method. I am also looking for some php samples on listening for these events as I am trying to create a Web GUI for a dialer that will allow me to show status of a call in real-time like Call In Progress, Call Ended, etc... I see that too many events are generated and I am wondering if there is an easy way of listening for a particular event? Would that be ActionID? if so, how to use it? Thanks a lot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] India Pune Pri call problem
India TRAI rules doesn't allow for CLID setting. They are backwards minded. If you ever get them to do it let me know ;) -Bruce On Mon, Feb 13, 2012 at 8:18 AM, Steven Howes steve-li...@geekinter.netwrote: On 13 Feb 2012, at 12:06, virendra bhati wrote: You can't set callerid for outgoing calls in case of PRI. Why not? Every PRI I have used supported it. Is this a carrier-specific thing? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?
Hi Sammy, Yes, that's what I have brain-stormed as well. I would very much appreciate sharing the code as I can try to improve it. I will put a GUI to it so that admin can insert campaign numbers and agents be able to see what percentage of the campaign is done. Can you share it with me already? Best, On Sun, Feb 12, 2012 at 12:50 AM, Sammy Govind govoi...@gmail.com wrote: Yes why not, I made an aut-odialer (the code I can share on my blogpost in couple of days for you.) The basic structure of the script/code was to: 1- Start, connect to DB, fetch campaign data 2- Fetch numbers to dial from campaign, If no numbers goto step 6 3- Feed those number in a loop to AMI using a php-AMI helper script (Async Event, don't wait for reply from Asterisk) 4- Check asterisk if its dialing capacity has reached or not 5a- If Not, goto step 2 5b- If Yes, wait for sometime for calls to finish, goto step 4 6- Close DB,Stop So, I had a context that was connecting to MySQL and on each incoming call trigger it was pushed with primary keys/identifiers of campaign and callednumber. Using those I updated the CDRs/STATUS of that particular number if it failed or successfully answered. That was all. Obviously there are major advanced features in this script which are missing and need time and proper coding expertise to develop..i.e multi-campaign mode, aggressiveness of dialer, retrying of failed numbers etc. Regards, Sammy. On Sat, Feb 11, 2012 at 9:23 PM, Bruce B bruceb...@gmail.com wrote: Sammy, Would you care to elaborate please. Have you had experience doing such a campaign using AMI? Maybe you can share of the code. Most appreciated, On Sat, Feb 11, 2012 at 10:15 AM, Sammy Govind govoi...@gmail.comwrote: I'd definitely go with AMI ! On Sat, Feb 11, 2012 at 6:39 PM, asterisk jobs asteriskcod...@gmail.com wrote: Thanks for the input but using spool files or AMI or AGI is way different from each other and that is what I want to get an input on. I do have hands on with all methods like I noted but want to know what the trend is now-a-days and what is more robust and proven out of all three. Best, On Sat, Feb 11, 2012 at 8:12 AM, David Backeberg dbackeb...@gmail.comwrote: On Sat, Feb 11, 2012 at 8:03 AM, asterisk jobs asteriskcod...@gmail.com wrote: Hi everyone, Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about 5000 numbers and then put the call to agents right away and pull up the CRM based on the number dialed. So, I am going to be doing some PHP+Ajax work. I am familiar with spool files but I don't like the fact that I can't read the status of the call in real-time. However, I know that it's the easiest way to approach the issue. The way to call 5000 numbers is to call one number, really well. Then you put it in a loop. You need to run a lab for long enough that you have the bugs worked out, before you subject real people to problems. With asterisk you can always tell the real-time status of a call, even if you initiate from a call file. Perhaps you would enjoy reading up on Local channels. Some people prefer to initiate calls from AMI. I tried it and didn't like it. But because most of us have been annoyed by an autodialer in our lives, even if we ourselves have made autodialers in the past, this is probably about the limit of the help you're going to get, unless you ask a more specific question that shows you've been trying to learn this hands-on and you've gotten stuck on a particular problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
