Re: [asterisk-users] How often to restart Asterisk...

2013-01-12 Thread Logan Bibby
I've actually had an AGI script that Asterisk never closed the fork for. It was testing a particular feature so it was pretty badly written. Ended up consuming a lot of resources. No idea why Asterisk hated that script, though. Failed to kill it every time. But would continue on the dial plan

Re: [asterisk-users] Verizon SIP trunking Field Trial

2013-01-05 Thread Logan Bibby
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 -- Best regards, Logan Logan Bibby, CEO Ke*o*bi Communications Tuscaloosa

Re: [asterisk-users] Timeout(absolute) not working on transfer

2012-12-30 Thread Logan Bibby
Geoff, I believe its actually TIMEOUT(absolute)=value. The function name is case sensitive. - Logan On Dec 30, 2012 9:53 AM, Geoff Lane ge...@gjctech.co.uk wrote: Hi All, Asterisk 1.4.22.1 on CentOS 5 I've configured my dialplan to limit the maximum call length on outgoing calls. I've

Re: [asterisk-users] Timeout(absolute) not working on transfer

2012-12-30 Thread Logan Bibby
No problem! Doubt check through a test extension. I don't want to be entirely wrong. ;) - Logan On Dec 30, 2012 12:12 PM, Geoff Lane ge...@gjctech.co.uk wrote: On Sunday, December 30, 2012, Logan Bibby wrote: I believe its actually TIMEOUT(absolute)=value. The function name is case

Re: [asterisk-users] Top Posting

2012-12-29 Thread Logan Bibby
I'm a +1 for the change, should it come to a vote. I realize the benefits of bottom-posting, especially when posting inline. But top-posting keeps things in reverse chronological order so any reader could catch up quickly on any missed messages in the chain. A new reader scrolls to the bottom and

Re: [asterisk-users] Top Posting

2012-12-29 Thread Logan Bibby
I suppose I'm one of the few people that remember the content of threads by subject and easily catch up... I'm also on my phone 99% of the time time and the way Gmail lays out emails makes top-posting beneficial to me. On Dec 29, 2012 8:57 PM, Richard Kenner ken...@gnat.com wrote: I realize

Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?

2012-12-27 Thread Logan Bibby
, Logan Logan Bibby, CEO Ke*o*bi Communications Tuscaloosa, Alabama -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] How to roll-over / move / rotate an Asterisk Master.csv call detail record (CDR) file every 15 minutes

2012-12-04 Thread Logan Bibby
I have a huge logrotate config file and I use Webmin to manage it all. Actually, Webmin is a good all-around system management tool, in my opinion. On Dec 4, 2012 9:12 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On 12-12-04 10:02 AM, Danny Nicholas wrote: IIRC log rotate only rolls

Re: [asterisk-users] How to roll-over / move / rotate an Asterisk Master.csv call detail record (CDR) file every 15 minutes

2012-12-04 Thread Logan Bibby
It is facing the outside world, but I just use SSH's port forwarding. :) On Dec 4, 2012 10:43 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Tuesday 04 December 2012, Logan Bibby wrote: I have a huge logrotate config file and I use Webmin to manage it all. Actually, Webmin

Re: [asterisk-users] Queue_log into MySQL - best practices

2012-11-22 Thread Logan Bibby
Have you considered using something like Splunk to aggregate your log files and store a copy for later analysis? Even if you want it to be available to someone, say a remote customer, via a web panel, I believe you could even have Splunk put it into another database or make a view in Splunk's

Re: [asterisk-users] Managing complex setups with Asterisk

2012-11-07 Thread Logan Bibby
What about just setting up a database which stores your data however you want then generate static files from that data or creating views for realtime (where appropriate)? That's how I do it with my company's system. To keep things not so complicated, I have AGI scripts. Keeps things clean and

Re: [asterisk-users] Bypass queue wrapup time

2012-10-29 Thread Logan Bibby
I don't think you can. But you could set it to a lower value like 3 seconds and give your operators a feature key to pause themselves in the queue if they need extra work time. - Logs On Oct 29, 2012 12:15 PM, Mitch Claborn mitch...@claborn.net wrote: In our sales queue, we have wrapup time set

Re: [asterisk-users] Fax for Asterisk success rates?

2012-10-04 Thread Logan Bibby
I had the same problem for a while. I found replacing fax machines with a scanner and either an email-to-fax program or just web-based faxing had better results. I don't want to tell you the gateway I used because they turned out pretty badly in the end. But there is hope! - Logan On Oct 4, 2012

Re: [asterisk-users] Reuse h extension?

2012-09-29 Thread Logan Bibby
I have a status context with a hangup extension. All my h calls go there. - Logan On Sep 29, 2012 4:32 AM, Stefan at WPF stefan.at@googlemail.com wrote: I have 2 contexts, however both have the same h extension. Currently I am doing copypaste for the h extension - is there a better way?

Re: [asterisk-users] Reuse h extension?

2012-09-29 Thread Logan Bibby
:-) 2012/9/29 Logan Bibby lo...@keobi.com I have a status context with a hangup extension. All my h calls go there. - Logan On Sep 29, 2012 4:32 AM, Stefan at WPF stefan.at@googlemail.com wrote: I have 2 contexts, however both have the same h extension. Currently I am doing

Re: [asterisk-users] Reuse h extension?

2012-09-29 Thread Logan Bibby
,hangup,2) ; - processes a channel not hung up by the dialplan On Sep 29, 2012 6:08 AM, Stefan at WPF stefan.at@googlemail.com wrote: Thanks Logan. Can you send an extract of your extensions.conf, how you do that? 2012/9/29 Logan Bibby lo...@keobi.com I do. I call the Hangup application

Re: [asterisk-users] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/

2012-09-27 Thread Logan Bibby
I agree. A script that read the spool directory, sent enough files to equal 10, wait a few seconds, check again and move more would do the trick. - Logan On Sep 27, 2012 11:27 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 09/28/2012 03:01 AM, Patrick Archibald wrote: Hi, Is

Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR

2012-09-25 Thread Logan Bibby
MyISAM would be best, in my opinion. The features that cause the little bit of performance overhead in InnoDB wouldn't be necessary for CDR storage. - Logan On Sep 25, 2012 4:15 PM, Matt Hamilton mistral9...@hotmail.com wrote: Which one (InnoDB or MyISAM) is preferred for CDR as far as write

Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR

2012-09-25 Thread Logan Bibby
Very good point. For revenue critical data like CDRs, being ACID compliant is important. MyISAM is compliant. And like InnoDB, can have the features making it compliant turned off. On Sep 25, 2012 6:12 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 09/25/2012 11:18 PM, Logan Bibby

Re: [asterisk-users] Peculiar problem with failover provision.

2012-09-24 Thread Logan Bibby
Why not use the DIALSTATUS channel variable to determine if a fail over is necessary? - Logan On Sep 24, 2012 6:00 AM, Thomas Kenyon dig...@sanguinarius.co.uk wrote: I have noticed a peculiar problem recently with the way that the failover operates in my dialplan. I normally have:

Re: [asterisk-users] Peculiar problem with failover provision.

2012-09-24 Thread Logan Bibby
I think a lot of people leave it out in examples for simplicity's sake. It doesn't instil proper practices in folks' heads. - Logan On Sep 24, 2012 12:06 PM, Eric Wieling ewiel...@nyigc.com wrote: You are doing it wrong. I know 50 bazillion Asterisk dialplan examples on the internet do it the

Re: [asterisk-users] How to get SIP Response Code and use it to change destination.

2012-09-23 Thread Logan Bibby
If you're using below 1.8, there isn't a way. The DIALSTATUS channel variable can give you a little, but not with those response codes. However, if you're using 1.8, there's some hope: you can use ${HASH(SIP_CAUSE,channel)} (where channel is the destination channel, not source) to read the SIP

Re: [asterisk-users] accept email and make phone call?

2012-09-20 Thread Logan Bibby
or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Logan Logan Bibby Ke*o*bi Communications Mobile: (205) 394-0424 -- _ -- Bandwidth and Colocation Provided by http://www.api