Re: [asterisk-users] AMOOCON debriefing
I was unable to join, but i promise i will be present one of the coming weeks. :) Zoa randulo wrote: Anyone who was at AMOOCON and who would deign to join us (ahem, Zoa, alors?) to hash out what happened and make fun of the presenters, please join us Friday at 6PM Paris time (5 PM UK) or 12 Noon EDT. I myself was really pleased to be there and meet so many interesting and amusing people. Some recorded discussion is also posted here: http://sessions.voipUsersConference.org Join us in a few minutes and every Friday for more about VoIP asterisk and the price of fish. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A side of Digium you may have never seen
I caught Mark Spencer, Kevin Fleming, John Todd, Russell Bryant, the other Mark in a truly Digium moment in Rostock, Germany on their way to listen to the sea shanties. http://tr.im/rawhide - be afraid, be very afraid (Adhearsions' Jason Goecke is also in the picture somewhere) /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMOOCON debriefing
Zoa, It was a pleasure to meet you! Please do come by some day, many people would like to talk about your work and your client! Does it do SIP URI? Call sip:7463#2262...@proxy.ideasip.com Best, Randy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incompatible changes to asterisk 1.6 MYSQL addon query syntax
I'm on asterisk 1.6.1.0 and asterisk addons 1.6.1.0 (also using freepbx
2.5 with cidlookup module from mysql database).
There are some incompatible changes in asterisk 1.6 about MYSQL addon
application syntax for querying a mysql database.
It seems that escaping of space and single quotes is no longer needed -
see 3rd line of attached example from
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL
I'd like to make sure the fix I submitted to freepbx is ok. Two
questions I cannot find answers to in the upgrade notes are:
- where the changes introduced from passing from 1.4 to 1.6 or where
they introduced during 1.6 branch?
- by experiment I saw that it's not necessary to quote spaces or single
quotes but can all escaping be removed? what still needs to be escaped
if anything?
exten = 888,1,MYSQL(Connect connid localhost ipcontact passwd ipcontact)
exten = 888,n,GotoIf($[${connid} = ]?error,1)
exten = 888,n,MYSQL(Query resultid ${connid} SELECT\ `number`\ FROM\
`phones`\ WHERE\ `channel`=\'${chan}\')
exten = 888,n(fetchrow),MYSQL(Fetch *foundRow* ${resultid} number) /;
fetch row/
exten = 888,n,*GotoIf($[${foundRow} = 1]?done)* /; leave loop if no
row found/
exten = 888,n,NoOp(${number})
exten = 888,n,*Goto(fetchrow)* /; continue loop if row found/
exten = 888,n(done),MYSQL(Clear ${resultid})
exten = 888,n,MYSQL(Disconnect ${connid})
thanks
John
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[asterisk-users] Rusting Snoms?
This is a bit off topic, because I 'think' it isn't an Asterisk problem. However I'm not sure and anyhow I'm hoping someone may recognize the symptom. We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years old) were packed up for the move, then unpacked a couple of weeks later. On unpacking them and connecting them to the new network, several of them didn't work well. The symptom is that outgoing RTP audio is garbled - like the packets are pulsed. Inbound is fine. This isn't true for all of the phones, just some of them. (The all run the same SNOM firmware) To be fair, they are on a new network, so it could be the cables or new 1Gb switches, except that the problem moves with the phone if you relocate it from one desk to another. I've tried a fresh asterisk install, but that didn't help either. So I am forced to conclude that something went 'bad' in those (old) phones while they were switched off. Has anyone got any clues for me? Thanks! Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rusting Snoms?
Because the phone is a digital system, I would suspect that it is a problem with the switch. Run a quick PCAP trace to see where the jitter comes from. Depending on the firmware version, you can do that from the web interface. CS -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Tim Panton Gesendet: Samstag, 9. Mai 2009 11:46 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Rusting Snoms? This is a bit off topic, because I 'think' it isn't an Asterisk problem. However I'm not sure and anyhow I'm hoping someone may recognize the symptom. We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years old) were packed up for the move, then unpacked a couple of weeks later. On unpacking them and connecting them to the new network, several of them didn't work well. The symptom is that outgoing RTP audio is garbled - like the packets are pulsed. Inbound is fine. This isn't true for all of the phones, just some of them. (The all run the same SNOM firmware) To be fair, they are on a new network, so it could be the cables or new 1Gb switches, except that the problem moves with the phone if you relocate it from one desk to another. I've tried a fresh asterisk install, but that didn't help either. So I am forced to conclude that something went 'bad' in those (old) phones while they were switched off. Has anyone got any clues for me? Thanks! Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over satellite internet
2009/5/9 Don E. Wisdom d...@engineeringinc.com I work on the salmon river in Idaho as a computer/radio tech. All of the satellite isp's do not have the upstream capability. Skype barely works. (you have to try upwards of 20 times for it to work) If I have to make phone calls when I am there I always use the SSB Radiophone or satellite phone because it is far far far more reliable and doesn't irritate the living hell out of the person your calling. I have tried 2-3 different VoIP providers all have the exact same result. The other side only hears a few pieces of word or nothing at all and hangs up. I have tried this on Starband (360 480 modems) wild blue Starband also has outages during the day where you cant see their satellite. Most of the satellite ISP's also have rolling bandwith caps. (Starbands is 1gig down 300megs up in a 7 day period for the plans I deal with) Overall I think its a bad idea. It most likely will not work. --Don Hello, i did once install in south Africa, and the only problem i had is the delay, however the client has the dedicated 2 mbit uplink. But when i talked over it the delay was really noticable. On 5/8/09 10:56 PM, Frank Bulk frnk...@iname.com wrote: If people don't mind taking turns talking, it will work. It's just going to be like talking on a CB. Reminds me of talking to my grandparents in the Europe as a child in the early 80's. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Fort Sent: Friday, May 08, 2009 10:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] VoIP over satellite internet Could those on the list who have used or tried to use VoIP over a satellite internet connection comment on how well it works or if it even works at all in a reliable way. What is the effect of latency on the VoIP path and how much is generally tolerable? routing via satellite adds about a quarter second of latency to the path. Is that too much? Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over satellite internet
Forgot to add, it is no so bad, i mean if you are in situation where local telco male you pay hell of a price. Or if you are in location not covered by any telco, i would go by sattelite option. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Professional Setup..
Not a taboo at all, you are providing your knowledge to setup the call center for example, and i your support in future. It is commen practice. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk blade server
Perfect office rackmount asterisk server? http://www.tgdaily.com/html_tmp/content-view-42372-135.html Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to run asterisk CLI commands from php
On Sat, 9 May 2009, Sam Hawkin wrote: I am trying to run the asterisk CLI commands from php. Some thing like asterisk -rx reload. $command = sudo asterisk -rx reload; $value1 = system($command,$retval1); Without any output or error messages its hard to guess... In what context is your PHP script executing? 1) As an AGI? 2) By you from a shell? 3) From cron? There are (at least) 4 elements to consider: 1) Does your script have the execute permission set? 2) Does the executing process have a clear path to your script? (Do all of the directories in the path to your script allow the executing process to execute your script.) 3) Is /usr/sbin/ in the path of the executing process? 4) Is the user running the script authorized to run sudo with no password? Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Professional Setup..
On Fri, 8 May 2009, Dave Walker wrote: I have a question for those who have done a few professional installs of Asterisk. Is it taboo to use something like AsteriskNow/FreePBX/Trixbox to get a base installation of Asterisk installed and functional for a small office? If not then do you always compile from scratch or use CentOS and the yum repositories? I used Asterisk At Home (predecessor of Trixbox) for my first couple of installs. I didn't need most of the cruft included and never took the time to understand all the funny little things that were done behind my back to make life easy for someone with little to no Linux skills. Fortunately, I was able to replace those systems before they were hacked by default passwords and buggy code. Now, I install a minimal CentOS (de-selecting every single package) and yum in just what I need. Then I install Zaptel, Libpri, and Asterisk from source. (I'm a 1.2 Luddite.) Parts left out don't get broke. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk blade server
2009/5/9 Dean Collins d...@cognation.net Perfect office rackmount asterisk server? http://www.tgdaily.com/html_tmp/content-view-42372-135.html Lacking dual hdd for raid 1. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Professional Setup..
2009/5/9 Steve Edwards asterisk@sedwards.com On Fri, 8 May 2009, Dave Walker wrote: I have a question for those who have done a few professional installs of Asterisk. Is it taboo to use something like AsteriskNow/FreePBX/Trixbox to get a base installation of Asterisk installed and functional for a small office? If not then do you always compile from scratch or use CentOS and the yum repositories? I used Asterisk At Home (predecessor of Trixbox) for my first couple of installs. I didn't need most of the cruft included and never took the time to understand all the funny little things that were done behind my back to make life easy for someone with little to no Linux skills. Fortunately, I was able to replace those systems before they were hacked by default passwords and buggy code. Now, I install a minimal CentOS (de-selecting every single package) and yum in just what I need. Then I install Zaptel, Libpri, and Asterisk from source. (I'm a 1.2 Luddite.) Parts left out don't get broke. Thanks in advance, Building everything from scrach each time is good when you have time, but when you want a nice web interface for cdr stats, end point management and good auto provisioning, it is not an option. Security issues are standing on good knowledge of the system and experience, to do not encounter default passwords and buggy code issues. Have fun. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Special Dialplan
Hello ppl,
I want to make a special dial plan for routing calls to a peer which
has an pin protection.
Normally if you want to call through that peer you must first enter
pin for example 1234#
and after that you hear the tone from line and after that you can dial
desired numbers.
I tried something like that, but doesn't worked. Did somebody have some clues?
exten = 0X.,n(dial1),Dial(SIP/peer-account/1234#${0x},15,rt)
Thank you guys for any help. I appreciate.
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Re: [asterisk-users] Special Dialplan
On Sat, May 9, 2009 at 4:22 PM, Catalin S. jonsonpla...@gmail.com wrote:
Hello ppl,
I want to make a special dial plan for routing calls to a peer which
has an pin protection.
Normally if you want to call through that peer you must first enter
pin for example 1234#
and after that you hear the tone from line and after that you can dial
desired numbers.
I tried something like that, but doesn't worked. Did somebody have some
clues?
exten = 0X.,n(dial1),Dial(SIP/peer-account/1234#${0x},15,rt)
Thank you guys for any help. I appreciate.
Start with an Answer() and then lose the r from your dial string. That
should allow you to press the code in.
If you want to hard code it, use dial and then probably Wait() followed by
senddtmf.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Special Dialplan
Thank you for your answer Steve,
Well I want to do this automatically... I mean if I want to route
0x through peer-account (which is pin protected), everything to be
automatically.
In fact i route through SIPURA SPA3102 Linksys fxo/fxs device , so my
device will answer and wait for my pin, if is ok wait for number to be
called.
Anyway did you know how can i send dtmf after is answered?
Thank you.
On Sat, May 9, 2009 at 11:45 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
On Sat, May 9, 2009 at 4:22 PM, Catalin S. jonsonpla...@gmail.com wrote:
Hello ppl,
I want to make a special dial plan for routing calls to a peer which
has an pin protection.
Normally if you want to call through that peer you must first enter
pin for example 1234#
and after that you hear the tone from line and after that you can dial
desired numbers.
I tried something like that, but doesn't worked. Did somebody have some
clues?
exten = 0X.,n(dial1),Dial(SIP/peer-account/1234#${0x},15,rt)
Thank you guys for any help. I appreciate.
Start with an Answer() and then lose the r from your dial string. That
should allow you to press the code in.
If you want to hard code it, use dial and then probably Wait() followed by
senddtmf.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Professional Setup..
Hmm words professional and trixbox don’t go together.. Did they actually fix the part where they put register - lines in between stanzas ? Trixbox , etc is nice to play around in your home , behind a firewall.. Anything that runs daemons like that, and vicidials shaddy little irc server.. is something to fear, i like to think they make it on purpose to make it hard to clone the systems they make , again, when it will break , you’ll need to fully understand the underlying spaghetti code , or be forced to tell your client to go buy trixbox support@ a year.. I personally HATE anything with trixbox in the name, but hey , it brings lots of paid support business.. Learn asterisk from scratch , you’ll save time in the long wrong.. You can make shell scripts to automate your installs, it’s probably faster than using a live cd that trixbox supply From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Walker Sent: May-09-09 1:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Professional Setup.. Greetings, I have a question for those who have done a few professional installs of Asterisk. Is it taboo to use something like AsteriskNow/FreePBX/Trixbox to get a base installation of Asterisk installed and functional for a small office? If not then do you always compile from scratch or use CentOS and the yum repositories? I setup FreePBX for a friend of mine who runs a small call center and I feel guilty profiting from something that so many other people have worked so hard to create. Ironically www.asterisknow.org has a 'Download - Free link but I don't see a 'donate' link.I wouldn't mind demanding my friend contribute some amount to help Asterisk grow but I'm not exactly sure how to do that and what the appropriate amount should be. I know Digium is a proud sponsor so I buy my phones, cards and refer people to their site if they need to buy equipment or an appliance. Has this topic come up for conversation in the past and if so then what was the outcome? Thanks! Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Professional Setup..
On Sat, May 9, 2009 at 1:56 AM, Dave Walker d...@damcconsulting.com wrote: Greetings, I have a question for those who have done a few professional installs of Asterisk. Is it taboo to use something like AsteriskNow/FreePBX/Trixbox to get a base installation of Asterisk installed and functional for a small office? If not then do you always compile from scratch or use CentOS and the yum repositories? I setup FreePBX for a friend of mine who runs a small call center and I feel guilty profiting from something that so many other people have worked so hard to create. Ironically www.asterisknow.org has a 'Download - Free link but I don't see a 'donate' link.I wouldn't mind demanding my friend contribute some amount to help Asterisk grow but I'm not exactly sure how to do that and what the appropriate amount should be. I know Digium is a proud sponsor so I buy my phones, cards and refer people to their site if they need to buy equipment or an appliance. Has this topic come up for conversation in the past and if so then what was the outcome? Thanks! Dave I like EVB (Easy Vox Box) much better than Trixbox. It doesn't have a the bloat. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Professional Setup..
I was not endorsing a particular product or asking for recommendations about specific products. I personally tried Trixbox and was not really satisfied with the results. The last time I was asked about an appliance I referred the person to Switchvox. There were a few other responses that seemed to answer my question. The vote tally so far (both on and off this list) seems to lean towards CentOS, download/compile of source then yum for the rest. For properly configured/secured small systems behind a firewall then FreePBX or AsteriskNow is fine. Nobody has commented on my question regarding monetary contributions for FreePBX, AsteriskNow or any other project that helps encourage support for Asterisk. I have contributed to the CentOS project. Their recommended contributions are $25 per installation, per year which is fine. Hmm words professional and trixbox don’t go together.. I have a question for those who have done a few professional installs of Asterisk. Is it taboo to use something like AsteriskNow/FreePBX/Trixbox to get a base installation of Asterisk installed and functional for a small office? If not then do you always compile from scratch or use CentOS and the yum repositories? I setup FreePBX for a friend of mine who runs a small call center and I feel guilty profiting from something that so many other people have worked so hard to create. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incompatible changes to asterisk 1.6 MYSQL addon query syntax
On Saturday 09 May 2009 03:42:04 John Fawcett wrote: I'm on asterisk 1.6.1.0 and asterisk addons 1.6.1.0 (also using freepbx 2.5 with cidlookup module from mysql database). There are some incompatible changes in asterisk 1.6 about MYSQL addon application syntax for querying a mysql database. It seems that escaping of space and single quotes is no longer needed - see 3rd line of attached example from http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL Remember the first time you had to do this and thought it was incredibly stupid how you had to escape commas and spaces? Well, nobody will think that anymore. What changed was in the process of making the comma the real separator, we made the insane amount of escaping go away completely. Yes, it's a one-time PITA that you have to change this, but it's wonderful for all of the people who come to Asterisk starting with 1.6 and never have to have seen the ugly amount of escaping. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Professional Setup..
On Sat, 9 May 2009, Dave Walker wrote: I was not endorsing a particular product or asking for recommendations about specific products. I personally tried Trixbox and was not really satisfied with the results. The last time I was asked about an appliance I referred the person to Switchvox. There were a few other responses that seemed to answer my question. The vote tally so far (both on and off this list) seems to lean towards CentOS, download/compile of source then yum for the rest. For properly configured/secured small systems behind a firewall then FreePBX or AsteriskNow is fine. Nobody has commented on my question regarding monetary contributions for FreePBX, AsteriskNow or any other project that helps encourage support for Asterisk. I have contributed to the CentOS project. Their recommended contributions are $25 per installation, per year which is fine. One way of reducing your guilt is by contributing - even if not actively maintaining the code, or contributing money, but by stay on the mailing list, answer questions, giving feedback to the maintainers, help others - that sort of thing. This goes for all open source projects though. So as for a professional setup... Well, I do it professionally... Well, I charge people money and for the most part they pay... I did look at the various pre-canned and GUI offerings when I started out, but I didn't really like them and they didn't seem too suited to what I thought at the time was my target market, so I started from scratch with my own thing. An advantage I had was that I already had a diskless booting Linux system that I was using to make routers and small NAS boxes from - boot off flash, run from ram, rather than run from flash, which most other systems seem to do... So I just had to compile (asterisk) from scratch, write my own dialplan, my own php front-end and off I went... I don't think there's any issue in you using Trixbox, etc. ... The main part, I think is actually knowing how to run a Linux server - that's something I think where most people might get stuck - it's all very well slapping in a CD or USB stick that loads up Linux, trixbox, pbxinnaflash, etc. but if you don't know much about how to actually care and feed the underlying operating system then I fear you'll come unstuck at some point. The advantage I have (In my view) is that I have a system exactly tailored to the underlying hardware. A custom Linux kernel compile (no modules, no udev, hoptplug or anything like that to get in the way), asterisk built for the platform (old i586 stuff - maybe not that relevant today!) and so on. There's no X windown GUI and nothing running that's not absolutely needed. Eg. one of my productions boxes: % ps ax | wc -l 34 And I've just looked at it and am wondering why there are 5 apache instances when one or 2 will do - so I can tweak the config file for the next release... And FWIW, that's running in 256MB of RAM, of which 132MB is taken up by a ramdisk, leaving the rest for applications. There is no swap. The ramdisk currently has 20MB free, but that's just fine. Maybe that's just me though - 28 years ago I was shoehorning assembly code into 200 bytes of RAM, so that sort of optimisation has stuck with me! Well, that's my tuppence of it all, anyway! Cheers, Gordon___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
