Re: [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals

Op 11-02-10 03:42, sean darcy schreef: > Kevin P. Fleming wrote: >> sean darcy wrote: >>> I found out that the [globals] section in extensions.conf is ignored if >>> an #include 'd file has a [globals] section. Is this intended? >>> >>> In this particular case, the #include 'd file has a number of

[asterisk-users] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)

Just to share some experience with everyone about what happened today to our Asterisk 1.4 box with Digium TE412P card. We had an unscheduled power outage which shut down the Asterisk box. When the power went up, Asterisk came back up okay but the ports on the card were all red. Zttool show red al

Re: [asterisk-users] Security Logging

Warren Selby wrote: > On Tue, Feb 9, 2010 at 5:54 PM, Lyle Giese > wrote: > > Here's a start for you, just run from cron once a day: > > Lyle > > > So basically, nothing built into asterisk that already provides > security logging mechanisms? Maybe I'm using t

Re: [asterisk-users] 1.6.2.1: DTMF trouble with PSTN

sean darcy wrote: > Tzafrir Cohen wrote: >> On Fri, Feb 05, 2010 at 01:55:03PM -0500, sean darcy wrote: >>> sean darcy wrote: Using 1.6.2.1 with a TDM400, attached to internal analog phones and PSTN. When I dial out to PSTN, I cannot send tones, like push "1" for something stupid.

Re: [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals

Kevin P. Fleming wrote: > sean darcy wrote: >> I found out that the [globals] section in extensions.conf is ignored if >> an #include 'd file has a [globals] section. Is this intended? >> >> In this particular case, the #include 'd file has a number of contexts >> for googlevoice. I'd put variou

[asterisk-users] Sending "Progress" during dialing

The PBX that I'm connecting to Asterisk has a timeout on calls on its PRI and QSIG lines. But that's smaller than the time it can take some SIP trunk providers to complete the calls, so I get hangups. I verified that sending "Progress" every few seconds will work around the problem. So I'd like

Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

On Thu, Feb 11, 2010 at 08:45:05AM +1300, Duncan Turnbull wrote: > The other way on Debian/Ubuntu is just to test the existence of the dir and > create it if needed > > If you add this to the /etc/init.d/asterisk near the start you should be fine > > if ! [ -d /var/run/asterisk ] ; then >

[asterisk-users] wellgate 3804A with frying

Dear Colleagues, I installed a Wellgate 3804A and overnight lines on all this with frying, putting other lines Wellgate 3804A is well, so I guess it's a problem the first team which is already out of warranty, anyone know how can I fix this? or where to send it in or capital Buenos Aires to fix

Re: [asterisk-users] IP Phone recommendation

At 02:18 PM 2/10/2010, you wrote: >I would love to hear some inputs on Aastra and Snom IP phones. I've have 3 480i-CT Aastra phones in our house for 3 or 4 years now with no complaints. Took a year for the firmware to get where it is and there were some things I'd like changed, but I can't remem

Re: [asterisk-users] How to avoid AGI script is canceled if callerHangUp

On Wednesday 10 February 2010 17:13:09 Steve Edwards wrote: > Un-top-posting... > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas > > Winter > > > > is there any way to avoid cancel the AGI script if caller is hanging up. > > That gives me sometimes data mismatch and it is

Re: [asterisk-users] How to avoid AGI script is canceled if callerHangUp

Un-top-posting... > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas > Winter > > is there any way to avoid cancel the AGI script if caller is hanging up. > That gives me sometimes data mismatch and it is deffcault to clean up in > the h extension. > > I would like that the

Re: [asterisk-users] How to avoid AGI script is canceled if callerHangUp

According to the CLI doc, you can do it this way - exten => 100,1,Set(AGISIGHUP=no) - exten => 100,n,AGI(youragi.agi) YMMV -- Danny Nicholas -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Winter Sent

[asterisk-users] How to avoid AGI script is canceled if caller HangUp

Hi, is there any way to avoid cancel the AGI script if caller is hanging up. That gives me sometimes data mismatch and it is deffcault to clean up in the h extension. I would like that the PHP script called by AGI will run to end.. Some thing can happend with an Macro if caller hang up exactly

Re: [asterisk-users] IP Phone recommendation

> I would love to hear some inputs on Aastra and Snom IP phones. I'm using Aastra 57i phones and like them. They can provisioned easily (without ANY entries from a local network). The support BLF and I'm also using the XML capability. --

Re: [asterisk-users] IP Phone recommendation

I would love to hear some inputs on Aastra and Snom IP phones. On Wed, Feb 10, 2010 at 4:36 PM, Jeff LaCoursiere wrote: > > On Wed, 10 Feb 2010, Tim Nelson wrote: > > > - "Gordon Henderson" > > > > wrote: > >> If not using PoE I'd suggest getting a few extra PSUs though - that's > >> one

Re: [asterisk-users] IP Phone recommendation

On Wed, Feb 10, 2010 at 3:21 PM, Philipp von Klitzing < klitz...@pool.informatik.rwth-aachen.de> wrote: > Here are two quotes that make me stay away from Cisco/Linksys: > > "Firmware can be downloaded from the Cisco Support Center (registered > partners only - password required) [...] > Here is a

Re: [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals

sean darcy wrote: > I found out that the [globals] section in extensions.conf is ignored if > an #include 'd file has a [globals] section. Is this intended? > > In this particular case, the #include 'd file has a number of contexts > for googlevoice. I'd put various googlevoice variables in the

Re: [asterisk-users] IP Phone recommendation

On Wed, 10 Feb 2010, Tim Nelson wrote: > - "Gordon Henderson" wrote: >> If not using PoE I'd suggest getting a few extra PSUs though - that's >> one >> area I have had a few issues with - but maybe it's just been the UK >> ones. >> >> Gordon > > The same can be said for the US versions. My e

Re: [asterisk-users] IP Phone recommendation

Hi! > Lack of features on the 7940's is frustrating, and makes me hesitant to > try other Cisco phones, even if the SPA504G is newer. Here are two quotes that make me stay away from Cisco/Linksys: "Firmware can be downloaded from the Cisco Support Center (registered partners only - password re

Re: [asterisk-users] IP Phone recommendation

- "Gordon Henderson" wrote: > If not using PoE I'd suggest getting a few extra PSUs though - that's > one > area I have had a few issues with - but maybe it's just been the UK > ones. > > Gordon The same can be said for the US versions. My experience has been it's not a case of 'if' the PS

Re: [asterisk-users] IP Phone recommendation

SPA504G is LINKSYS with newer look and HD :-) Expect all you had in Linksys SPA9XX + more. I personaly have both phones - differences are not a lot :) Peter On 10.2.2010 20:31, Jeffrey Ollie wrote: > On Wed, Feb 10, 2010 at 12:23 PM, Brent Torrenga wrote: >> >> Coming from someone who uses

Re: [asterisk-users] IP Phone recommendation

On Wed, 10 Feb 2010, Sebastian Milioto wrote: > Hi all, > > I have to install 25 IP Phone in some building. I want just basic IP Phones > like: > > > Cisco-Linksys SPA922 u$s 146 > Grandstream GXP-2000 u$s 105 > Snom 300 u$s 119 > > The most valuables param

Re: [asterisk-users] billing based on local access number

At 4:02 AM on 10 Feb 2010, umesh maharjan wrote: > > Hi all, > > I am configuring asterisk as a prepaid calling card. I am getting > different local rate from my ISDN provider e.g 0.002 for landline > and 0.13 for mobile etc. In this case I thing I have to say my > asterisk/a2billing to bill b

Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

The other way on Debian/Ubuntu is just to test the existence of the dir and create it if needed If you add this to the /etc/init.d/asterisk near the start you should be fine if ! [ -d /var/run/asterisk ] ; then mkdir /var/run/asterisk chown $AST_USER.$AST_GROUP /var/run/asterisk

[asterisk-users] problems with 1.6

In an attempt to fix problems with EAGI delays in 1.4 (see my other message for more on that), I've tried upgrading to 1.6, in case it's a bug that's fixed in the newer version. Unfortunately, I'm having all kinds of trouble with this new install. My system relies on conferences, and whenever I

Re: [asterisk-users] IP Phone recommendation

On Wed, Feb 10, 2010 at 12:23 PM, Brent Torrenga wrote: > Coming from someone who uses 7940's and 60's: has Cisco/Linksys embraced > SIP compatibility with asterisk more completely with the SPA504G's than > they > have the 7940 series? Lack of features on the 7940's is frustrating, and > makes

Re: [asterisk-users] How to run a remote PHP script and still have access to audio stream?

Hello David, Thanks so much for your message! Please check my comments inline below... David Backeberg wrote: > On Sun, Feb 7, 2010 at 9:54 PM, Leo Burd wrote: > >> Hello there, >> >> I'm trying to figure out how to run a PHP script on a remote machine and >> still have access to the audio

Re: [asterisk-users] Nat Issue - is this Draytek || Asterisk?

On Wed, Feb 10, 2010 at 5:53 AM, Brian < brel.astersik100...@copperproductions.co.uk> wrote: > I'm struggling to work out how can I debug this effectively and would > appreciate some guidance here. > > Try enabling sip debug on the internal peers (sip set debug peer from the cli) before you b

Re: [asterisk-users] How to run a remote PHP script and still have access to audio stream?

On Wed, 2010-02-10 at 13:47 -0500, Leo Burd wrote: > Is there any > easy way for Asterisk to play audio files located in remote servers? If you can mount it, Asterisk will happily read from it. Perhaps you can run a an ssh/ftp/smb/nfs server deamon on the webserver and mount it on the filesyste

Re: [asterisk-users] Security Logging

On Tue, Feb 9, 2010 at 5:54 PM, Lyle Giese wrote: > Here's a start for you, just run from cron once a day: > > Lyle > So basically, nothing built into asterisk that already provides security logging mechanisms? Maybe I'm using the wrong term; In Windows, I think it would be called Security Audi

Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

On Wed, 2010-02-10 at 11:24 -0600, Jason Parker wrote: > Brian wrote: > > Each time the server is rebooted Asterisk duly > > deletes the manually created /var/run/asterisk directory - quite why it > > does this I just don't know - perhaps it is a bug? > > > > Your assumption is incorrect. Some L

Re: [asterisk-users] How to run a remote PHP script and still have access to audio stream?

Hello Ben, thanks for your message! I'm implementing a framework to integrate Asterisk and Drupal (a powerful tool for the creation of social- and media-rich websites). Since the voip and the web components of the system are likely to run on separate servers owned by different organizations, I

Re: [asterisk-users] IP Phone recommendation

On Wed, Feb 10, 2010 at 12:23 PM, Brent Torrenga wrote: > > Coming from someone who uses 7940's and 60's:  has Cisco/Linksys embraced > SIP compatibility with asterisk more completely with the SPA504G's than they > have the 7940 series?  Lack of features on the 7940's is frustrating, and > makes m

Re: [asterisk-users] IP Phone recommendation

>SPA504G - 1 more vote for it. > >It is worth having 4 lines even if you need 1 initially. > >SPA504G supports G722 and sound is awesome even if you do not not use >teh HD sound. If you do not care that mcuh about HD sound and do not >need PoE SPA941 is a excellent choice - you get really a lot f

Re: [asterisk-users] IP Phone recommendation

I see... problem with spa941 is it dont have LAN port (I'm thinking NAT the customer PC) Sebastian On Wed, Feb 10, 2010 at 2:10 PM, Peter wrote: > On 10.2.2010 18:06, Steve Howes wrote: > > On 10 Feb 2010, at 15:50, Peder wrote: > >> check out the Cisco SPA504G. They are the newer versions

Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

Brian wrote: > Each time the server is rebooted Asterisk duly > deletes the manually created /var/run/asterisk directory - quite why it > does this I just don't know - perhaps it is a bug? > Your assumption is incorrect. Some Linux distributions will empty /var/run/ on boot, just as they do wit

[asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals

I found out that the [globals] section in extensions.conf is ignored if an #include 'd file has a [globals] section. Is this intended? In this particular case, the #include 'd file has a number of contexts for googlevoice. I'd put various googlevoice variables in there to use in all those cont

Re: [asterisk-users] Muted calls occasionally dropping after 30 seconds

Ishfaq- > I'm having a very odd phenomenon happening on our production server > (1.4.17 and using realtime). Sometimes a call will disconnect 30 seconds > after the SIP phone hits the mute button but it doesn't happen all the > time. I've done a sip debug while watching this happen and that doesn'

Re: [asterisk-users] IP Phone recommendation

On 10.2.2010 18:06, Steve Howes wrote: > On 10 Feb 2010, at 15:50, Peder wrote: >> check out the Cisco SPA504G. They are the newer versions of the >> SPA922, support multiple lines and are fairly cheap too. > > I second that. They're rock solid, good audio quality and easy to > provision. >

[asterisk-users] asterisk sudden restart - 1.4.18.1

Hi, Asterisk got stopped this morning after 20 minutes and phones went to 'No Service' and then got started automatically after 20 min, as I could see in the full log that asterisk got started at so and so time: [Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Event Logger Started /var/log/aste

Re: [asterisk-users] IP Phone recommendation

-BEGIN PGP SIGNED MESSAGE- Hash: SHA512 I'd like to jump in here as well, with the Aastra 57i. It is easy to configure with asterisk, provision and is not that badly priced either. - - Tommy William Stillwell (Lists) skrev: > Polycom 331’s are also in the same price range, and offer good

Re: [asterisk-users] IP Phone recommendation

Polycom 331's are also in the same price range, and offer good features as well. All my polycoms are provisions with option 66 on dhcp, and an ftp site with cfg files that are build from a mysql database from sip users table. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-us

[asterisk-users] EAGI delay

Hello, I made a post to the forums (http://forums.digium.com/viewtopic.php?f=1&t=72901&sid=3d5c2717ca5ab7ad676957ae436d4b51) but haven't received any replies, so thought I'd try here. On my debian machine running asterisk 1:1.4.21.2~dfsg-3, I've been noticing that there's a problem with confe

Re: [asterisk-users] IP Phone recommendation

On 10 Feb 2010, at 15:50, Peder wrote: > check out the Cisco SPA504G. They are the newer versions of the > SPA922, support multiple lines and are fairly cheap too. I second that. They're rock solid, good audio quality and easy to provision. S --

Re: [asterisk-users] IP Phone recommendation

I was recommended Polycom phones. I tested some. And now, I LOVE them. Look at the Polycom IP321. It's a great phone with provisioning and two lines. Dont know about G729, but I'd be surprised if it didn't support it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun..

Re: [asterisk-users] Still on spandsp/app_fax and T.38

> > - "Kevin P. Fleming" escreveu: > > > Vinícius Fontes wrote: > > > Will do. You guys will have my feedback on monday. If everything > > goes okay with that change, I'll post a patch on Mantis. > > > > No need for the patch; it's already on my radar, and if you confirm > > that > > it ac

Re: [asterisk-users] IP Phone recommendation

Don't use Grandstream if you want quality and stability. Also check out the Cisco SPA504G. They are the newer versions of the SPA922, support multiple lines and are fairly cheap too. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Se

Re: [asterisk-users] Optimization of call from server 1 to 2 and thenback to 1

Hi Danny, sorry you are correct: > Difficult to say since you don't say if you are on 1.2, 1.4 or 1.6 both Asterisk are running version 1.4.21.2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asteri

Re: [asterisk-users] Know what would be killer?

On Thu, Feb 4, 2010 at 7:39 PM, Lyle Underwood wrote: > If call recordings were stored in stereo and the callers were evenly > distributed along the stereo spectrum. BAM. Cisco has this. It's called telepresence. It costs a LOT of money, and takes a LOT of bandwidth, but you do get spatial distri

Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

On Wed, Feb 10, 2010 at 5:23 AM, RESEARCH wrote: >> >> snip >>> >>> You are correct. 8 span which process up to 240 calls at pick time >>> If the system is actually performing fine then I'd just say that there >>> is something about the Asterisk threads that makes them look runnable >>>

[asterisk-users] IP Phone recommendation

Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stabi

Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

On Wed, 2010-02-10 at 14:14 +, Brian wrote: > On Wed, 2010-02-10 at 08:54 -0500, Ken Leland III wrote: > > Brian, > > > > It could be that the ownership/permissions on the directory are not correct. > > Are you running asterisk as asterisk:asterisk or root:root? > > > > Here is an article tha

Re: [asterisk-users] problems with creating a call

hehe i figured it out.. it was really stupid :) i use opensips as an sip proxy, and i configured opensips to only react on packages from local ip's.. asterisk was sending to an external ip and that way i created my own little loop :), changed in sip.conf all the hosts to the internal ip of opensip

Re: [asterisk-users] Optimization of call from server 1 to 2 and thenback to 1

Difficult to say since you don't say if you are on 1.2, 1.4 or 1.6, but my WAG would be that the IAX connection takes this out Asterisk 1's hands. The attendant transfer never breaks the IAX connection; it actually creates an extra IAX connection to let A talk to C like this: Original call A --> I

[asterisk-users] Optimization of call from server 1 to 2 and then back to 1

Hi All, suppose this call flow: there are two Asterisk servers, they are connected through a IAX2 trunk. The users use SIP. The user A on the Asterisk server 1 calls the user B on the Asterisk server 2. They talk for a while and then the user B does an attendant transfer to the user C on the

Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

It seems to me that the restart is creating asterisk.pid in the "wrong" place. Try this - - find /|grep asterisk.pid This will tell you where the mislocated pid is being created and you can adjust the script accordingly. -- Danny Nicholas -- -Original Message- From: asterisk-users-

Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

On Wed, 2010-02-10 at 08:54 -0500, Ken Leland III wrote: > Brian, > > It could be that the ownership/permissions on the directory are not correct. > Are you running asterisk as asterisk:asterisk or root:root? > > Here is an article that lists the directories and what the > ownership/permissions

[asterisk-users] Muted calls occasionally dropping after 30 seconds

Hi I'm having a very odd phenomenon happening on our production server (1.4.17 and using realtime). Sometimes a call will disconnect 30 seconds after the SIP phone hits the mute button but it doesn't happen all the time. I've done a sip debug while watching this happen and that doesn't show an

Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

Brian, It could be that the ownership/permissions on the directory are not correct. Are you running asterisk as asterisk:asterisk or root:root? Here is an article that lists the directories and what the ownership/permissions on each one should be: http://www.voip-info.org/wiki/view/Asterisk+non

Re: [asterisk-users] problems with creating a call

Peter den Hartog wrote: > Hello, > > I installed Asterisk in a linonde cloud debian 5, and i'm trying to > create a first call but when i try to set up the call i see the > following message: > > -- Called 1...@100 > -- Now forwarding SIP/105-0008 to 'Local/1...@default' (thanks to >

[asterisk-users] problems with creating a call

Hello, I installed Asterisk in a linonde cloud debian 5, and i'm trying to create a first call but when i try to set up the call i see the following message: -- Called 1...@100 -- Now forwarding SIP/105-0008 to 'Local/1...@default' (thanks to SIP/100-0009) -- Executing [...@de

Re: [asterisk-users] forward incomming line to modem

On Wed, 2010-02-10 at 12:02 +0200, Tzafrir Cohen wrote: > On Wed, Feb 10, 2010 at 07:52:06AM +0100, randall wrote: > > hi All, > > > > its probably very simple but i can't find the way to it. > > > > i have some b410p cards and use them to connect to ISDN2, this works OK > > for calling but i nee

[asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

Since upgrading from 1.6.1 to 1.6.2 I get this error on boot: Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory Or if I try to connect to Asterisk: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) If I manually create /var/run/ast

[asterisk-users] Nat Issue - is this Draytek || Asterisk?

I'm trying to debug a NAT issue and I can't make up my mind if the problem is with my Vigor 2800 or Asterisk 1.6.2. I know the Draytek is alleged to suffer from nat 'issues' but I did not have the issue with 1.6.1 - so I'm wondering if something has changed? The Draytek offers 'NAT & Routed' on a

Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

On Wed, Feb 10, 2010 at 01:23:01PM +0300, RESEARCH wrote: > @Steve Edward. Can you share your C agi codes? I presume what you want me to > do is rewrite the script in C and use it as compiled binary Yes. But then again, for such a simple call (a single INSERT) you can use a MySQL() from the dial

Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

> > snip >>> >> >> You are correct. 8 span which process up to 240 calls at pick time >> >>> If the system is actually performing fine then I'd just say that there >> is something about the Asterisk threads that makes them look runnable >> and that >>> accounts for the high load average. ?Is the I

Re: [asterisk-users] forward incomming line to modem

On Wed, Feb 10, 2010 at 07:52:06AM +0100, randall wrote: > hi All, > > its probably very simple but i can't find the way to it. > > i have some b410p cards and use them to connect to ISDN2, this works OK > for calling but i need to have 1 line to be send to the fax machine. > BRI fax machine?

Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

On Wed, Feb 10, 2010 at 04:26:48AM -0500, Steve Totaro wrote: [snip] > > my $data0= $ARGV[0]; > > my $data1= $ARGV[1]; > > my $data2= $ARGV[2]; > > my $data3= $ARGV[3]; > > my $data4= $ARGV[4]; > > my $data5= $ARGV[5]; > > my $data6= $ARGV[6]; > > my $data7= $ARGV[7]; [snip] > > > > my $insert_

[asterisk-users] PMS (SDMR, ...) support in Asterisk

Hello, In this list archives, you can find here and there threads related to Property Management System support in Asterisk. Google shows this doc (http://www.mitel.com/resources/guide_8922_misn.pdf) which gives an interesting overview of this topic. 1. Is this Station Message Detail Recording wi

Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

On Wed, Feb 10, 2010 at 10:12:55AM +0300, Muro, Sam wrote: > > >> Hi Team > >> > >> Can someone advice me on how i can lower the load average on my asterisk > >> server? > >> > >> dahdi-linux-2.1.0.4 > >> dahdi-tools-2.1.0.2 > >> libpri-1.4.10.1 > >> asterisk-1.4.25.1 > >> > >> 2 X TE412P Digium c

Re: [asterisk-users] asterisk and mysql connection

김무성 wrote: > > Hell list. > > I wanna use mysql for storing user’s ID or etc. > > If user call to other, asterisk have to search number in mysql. > > Are there document about setting asterisk and mysql? > > THX > > Kim > Hi You need to read up on Asterisk Realtime http://www.voip-info.org/wiki/vi

Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

snip >> > > You are correct. 8 span which process up to 240 calls at pick time > >> If the system is actually performing fine then I'd just say that there > is something about the Asterisk threads that makes them look runnable > and that >> accounts for the high load average.  Is the IVR an agi or

Re: [asterisk-users] asterisk and mysql connection

- "김무성" wrote: Hell list. I wanna use mysql for storing user’s ID or etc. If user call to other, asterisk have to search number in mysql. Are there document about setting asterisk and mysql? http://www.voip-info.org/wiki/view/Asterisk+RealTime -- Thanks, Phil -- __

[asterisk-users] asterisk and mysql connection

Hell list. I wanna use mysql for storing user’s ID or etc. If user call to other, asterisk have to search number in mysql. Are there document about setting asterisk and mysql? THX Kim -- _ -- Bandwidth and Colocation P

Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

- "Sam Muro" wrote: > >> Hi Team > >> > >> Can someone advice me on how i can lower the load average on my > asterisk server? > >> > >> dahdi-linux-2.1.0.4 > >> dahdi-tools-2.1.0.2 > >> libpri-1.4.10.1 > >> asterisk-1.4.25.1 > >> > >> 2 X TE412P Digium cards on ISDN PRI > >> > >> Im using the

Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

On Wed, 10 Feb 2010, Muro, Sam wrote: [snip] > I have the agi scripts not as ivr but to help populate the required > information into mysql db. Probably here is where the problem lies i > have to connect and disconnect to mysql each time a call is made or a > specific menu is selected > > Here