Re: [asterisk-users] AGI - Choppy Sound
Sometimes my process just drops the call and leaves the client in silence! it happens probably 1 out of 25 times... no meaningful loads happening, just me prototyping my (Ruby)AGI... On Wed, May 7, 2008 at 12:40 PM, Robert Norton - SophTelecom LLC [EMAIL PROTECTED] wrote: Hi Marcelo, Sorry, just realized I responded to you directly rather than the list… So for the record, here's the list response… -- Hi Marcelo, What format are the recordings in? Have you tried converting them to the same format? Thanks -- * * *From:* Marcelo Freitas [mailto:[EMAIL PROTECTED] *Sent:* Monday, May 05, 2008 6:53 PM *To:* Robert Norton - SophTelecom LLC; 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* RE: [asterisk-users] AGI - Choppy Sound Hi Robert, Thanks for replying and I'm glad you have an application running nicely through phpAGI ... Now answering your questions ... what's the load on the box during the times it's choppy? I was testing at night ... I mean NO simultaneous calls going through the server ... It is good machine (Dell Xeon 2.8GHZ, 1GB RAM ... cpu load 10%) ... I also notice the problem as other calls were up too ... sometimes the quality is bad ... sometimes good Are calls in general choppy during that same point or just calls going through AGI? I never had this problem with my normal menu ... and when I call and I have the problem I tried to hangup and call the other number for the other menu ... and ... no problems ... It's hard to verify ... because I call ... it's choppy ... hangup and call again ... sometimes it's choppy sometimes not ... You mention the attendant voice becomes choppy? Yes, when I call and the call is not good, and during the same call I try to talk to one atendant, I almost cannot hear her/his, but the sound for him/her is good ... That's why I don't know if it is a problem with the recordings I did, because the agent's voice is also bad Is the attendant totally outside of your AGI scripts? I'm sorry, what did you mean ? Usually what I do is ... Answer the incoming call - send to AGI - it does the logic and play some sounds - and I do and exec_goto to an context,extension,priority that has a queue setup - and from there on they answer the calls What codecs are your clients using? The incoming calls - IAX/ilbc Connection to agents - SIP/ulaw it's the same as the other menu ... Thanks, - Original Message - Subject: RE: [asterisk-users] AGI - Choppy Sound From: Robert Norton - SophTelecom LLC [EMAIL PROTECTED] Date: Mon, May 5, 2008 19:47 Hi, I take it you've looked at all the basics, what's the load on the box during the times it's choppy? Are calls in general choppy during that same point or just calls going through AGI? You mention the attendant voice becomes choppy? Is the attendant totally outside of your AGI scripts? What codecs are your clients using? I'm working on a pretty intensive phpAGI based application and even with a decent number of calls haven't had any substantial problems, more so just with load but even with substantial activity on a fairly robust box it has been fine. Thanks -Robert Norton -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Marcelo Freitas *Sent:* Monday, May 05, 2008 4:10 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] AGI - Choppy Sound Hi folks, I'm experiencing some problems with sound through phpAGI ... What I'm trying to do is a menu, doing some database lookups and so ... But sometimes the sound become too choppy ... just sometimes .. like 1 of 5 calls ... but is a big percentage ... And I have my current menu on the dialplan that I have no problems with it ... I'm using .gsm for both but different recordings ... Does anybody has had problems like that ? Is it AGI performance problem ... even the atendant voice becomes choppy ... So strange ... Does anybody have a recommendation ? Thanks, ___ -- 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 You can tell whether a man is clever by his answers. You can tell whether a man is wise by his questions - Naguib Mahfouz ___ -- 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] When calling in via AGI, gsm sound file plays but sometimes drops out
Hi. I am using the 'get_data' function from an AGI, and i find that sometimes when users call in, it won't play the full gsm soundfile, and when i try to press a number (or pound, or star), nothing will happen - it just hangs there... anyone else experience this? - Dominic Son It is not the force of a stroke that makes fine art ___ --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] Does Anyone Have a StanaPhone Number here?
Could you please call it and confirm with me it's not working for you either? I should probably transfer my DID number anyways, if I could only get them to respond! Does anyone have a suggestion as to where to go in this situation? Possibly a place with high capacity concurrent incoming calls... -- Anything else, let me know. - Dominic It is not the force of a stroke that makes fine art ___ --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] In my messages log..
Hi. I'm still a bit of a newb to linux, I see this in my messages log: Asterisk init: Id ax respawning too fast: disabled for 5 minutes What does this mean? and how severe is it? -- Anything else, let me know. - Dominic It is not the force of a stroke that makes fine art ___ --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.conf and it's impact on CLI
I was previous using Asterisk 1.2.9.1 and decided to get some real servers outside of my house. It was time for Asterisk 1.4.4. I figured since all the conf files were in /etc/asterisk form the old box, i'd just copy tha directory over to the new server. My SIP DID AGI stuff worked, except running 'asterisk -r' doesn't. It tells me ' Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)' Basically, the difference between 'asterisk.conf' file is as follows: v 1.2.9 (installed through trixbox) astrundir = /var/run/asterisk v 1.4.4 astrundir = /var/run So in my new servers, if i keep it as '/var/run/asterisk, my DID phone will work with stanaphone (in which i'm crapping in my pants if they'll exist cause they never return emails). Though CLI won't work. if i do '/var/run', my DID won't work, but CLI will... I've tried just coping over the extensions_additional.conf and sip_additional.conf files from my old setup to my new one, and that didn't work. Maybe I should just install my previous version. Are there QoS differences though? I'd rather not regress if that were the case. -- Anything else, let me know. - Dominic It is not the force of a stroke that makes fine art ___ --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.conf and it's impact on CLI
awesome. it worked. thanks guys. On 10/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Oct 20, 2007 at 11:57:05PM +0530, Jaswinder Singh wrote: astrundir = /var/run Change this to astrundir = /var/run/asterisk on 1.4 server and chmod /var/run/asterisk to 777 . make sure u create that directory as well . chmod 777 (or even 666) to the control socket (asterisk.ctl) allows everybody on the system to write to it, and hence have full control of Asterisk. In short: bad. You fix things with chown, not with chmod 666/777 . And anyway, for it to have a lasting effect you need to set up the permissions in asterisk.conf anyway. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Question about the Trixbox and GSM optimization
Hi. In the Asterisk TFOT book, under installing Asterisk, it says as an option: Uncomment the following line in your Asterisk Makefile to enable GSM codec optimizations on x86 CPU architectures that support MMX instructions: #K6OPT = -DK6OPT Does installing from the basic Trixbox CD possibly do something like check what kind of CPU you have, and if you quality, turns this option on? -- Anything else, let me know. - Dominic ___ --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] AGI with System() ?
Ok, this is what worked: EXEC System rm -rf /var/lib/asterisk/sounds/blah.gsm the -rf eliminates the hassle.. a dream come true it worked ! On 10/13/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Oct 13, 2007 at 05:45:26PM -0700, Dominic Son wrote: Hi. You mean to use the AGI funtion in the particular programming language? yeah. i tried, same results.. : T I guess that this is a permissions issue. Check what you get in the standard error. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] AGI with System() ?
Uuugh..for the life of me, i cannot delete sound files using
EXEC System(rm /var/lib/asterisk/sounds/blah.gsm)
through AGI
the AGI debug log indicates the command executes successful ( equals 0)
but my files are clearly still there.
If i try System(rm ...) in my extensions.conf diaplan it'll work there.
Is there a bug in the AGI to use System ?
because i tried to copy files ('cp') as a test, and that didn't work either..
I'm running Trixbox v 1.1.0
Asterisk 1.2.9.1 svn rev 34876
with RAGI
--
Anything else, let me know.
- Dominic
___
--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] AGI with System() ?
tried both as suggested...though AGI says it's succesful: AGI Rx EXEC System rm /var/lib/asterisk/sounds/abandons.gsm AGI Tx 200 result=0 the abandons.gsm file is still there... i have to delete it through my agi because i'm recording sounds, and i want users to hear their recording and redo it if they choose to. so there's a 'RECORD FILE', but not a 'DELETE FILE' .. : T On 10/13/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Dominic Son wrote: Uuugh..for the life of me, i cannot delete sound files using EXEC System(rm /var/lib/asterisk/sounds/blah.gsm) through AGI agi show exec Usage: EXEC application options Executes application with given options. So I'd try EXEC System rm foo or EXEC System rm\ foo But I don't understand why you would want to do that instead of just running the command in your script. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] AGI with System() ?
Hi.
You mean to use the AGI funtion in the particular programming
language? yeah. i tried, same results.. : T
i guess i'll have to put it in a database, and flag it to remove
manually for now...
On 10/13/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Dominic Son wrote:
tried both as suggested...though AGI says it's succesful:
AGI Rx EXEC System rm /var/lib/asterisk/sounds/abandons.gsm
AGI Tx 200 result=0
the abandons.gsm file is still there...
Umm, then I don't know what's going wrong.
i have to delete it through my agi because i'm recording sounds, and i
want users to hear their recording and redo it if they choose to.
so there's a 'RECORD FILE', but not a 'DELETE FILE' .. : T
On 10/13/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Dominic Son wrote:
Uuugh..for the life of me, i cannot delete sound files using
EXEC System(rm /var/lib/asterisk/sounds/blah.gsm)
But I don't understand why you would want to do that instead of
just running the command in your script.
You did not get my point. Why not just do something like
unlink('/var/lib/asterisk/sounds/abandons.gsm');
or
exec('rm -f '.escapeShellArg('/var/lib/asterisk/sounds/abandons.gsm'));
in your script? (examples are for PHP)
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
___
--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] Exposing sound files through http for links
Hi. I'd like for my sound files to be exposed through http. You know, the ones located in var/lib/asterisk/sounds. This is probably an apache thing i'd have to configure or is accessible through some asterisk http routing? 1. how one would configure this? 2. what are the security costs of doing this to asterisk? - Dominic ___ --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 on NGINX Server?
Hi. Was the AGI Server to write dialplans in any programming language in Asterisk assumed to be configured for the apache web server? Or should it not matter what web server you have (in my case NGINX)? - Dominic The ability to simplify means to eliminate the unnecessary so that the necessary may speak. -Hofstadter's Law ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Abstraction for a newbie
Thank you Mark. I've went from The number you are dialing is not in service, please check the number and dial again to a fast busy tone...I think I'm getting closer..-- Anything else, let me know. -Dominic Sonwww.DominicSon.comOn 8/12/06, Mark Phillips [EMAIL PROTECTED] wrote:Sounds to me like you don't have a proper connection with Stanaphone. The only time you'll get these problems is when they cannot contact youto forward the call to your system.Double check you firewall settings. They need to be able to reach yoursystem on port 5060UDP (assuming SIP) as well as ports 1-2UDP (Asterisk default media ports).They'll contact yo when a call comes in. You'll accept the call and atthe same time tell them which port to send the incoming audio to.They'll also tell you where to send your outgoing audio. Hope that helps.MarkOn Fri, 2006-08-11 at 15:45 -0700, Dominic Son wrote: Hi. Can someone explain to a right brained person what is going on with In/out bound trunks, how it connects to my Trixbox.. 1. i get issued a free NY phone number from a voip service like stanaphone . 2. i then call this number, it connects to the stanaphone voicemail 3. i turn off the voicemail because i want it to connect to my Askterisk, I've set up all the trunks in the PBX setup, ( sip.stanaphone, etc) 4. now i call my NY number, and it says 'this phone is not in service, please check the number and dial again' my Q: how does this work, more specifically, if i turned off the VM, how does stanaphone then know to look for my asterisk server to use the trixbox? -- Anything else, let me know. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Abstraction for a newbie
Hi. Can someone explain to a right brained person what is going on with In/out bound trunks, how it connects to my Trixbox..1. i get issued a free NY phone number from a voip service like stanaphone . 2. i then call this number, it connects to the stanaphone voicemail 3. i turn off the voicemail because i want it to connect to my Askterisk, I've set up all the trunks in the PBX setup, ( sip.stanaphone, etc)4. now i call my NY number, and it says 'this phone is not in service, please check the number and dial again' my Q: how does this work, more specifically, if i turned off the VM, how does stanaphone then know to look for my asterisk server to use the trixbox?-- Anything else, let me know. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with newbie: D-link admin setup
Hi. can someone help me with where in the D-Link menu I'd go about configuring it to allow people to access my internal Asterisk server.I just want to make a call to this stanaphone number and see something happen on Asterisk. I have a Dinky DI-524 Thanks very much.-- Anything else, let me know.-Dominic Son ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
