similar to: Script to softly restart Asterisk each midnight to clean locked channels

Hello, Anybody can advice how to pass variable between 2 Asterisk servers over IAX2? With SIP I can use SipAddHeader. How do to the same with IAX2? Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com

Hello, I have configured one provider in Asterisk Realtime DB without username and password, only host=<providers_IP> and ipaddress=<providers_IP> Now when I'm trying to send call using this provider I'm using following string: Dial(SIP/NUMBER at Provider) In Asterisk 1.4.15 debug I see that Realtime engine is using query: [Dec 20 00:02:15] DEBUG[14634]:

Hi I have some question about mysql realtime. 1) Anyone know exactly if there is a specific order to declare sip table column for realtime ? In which file can I find that order ? 2) In my extconfig.conf, [settings] are : sipusers => mysql,general,siptable sippeers => mysql,general,siptable so means that I use realtime dynamic exactly ? Is it normal if some parameters from sip.conf still

Hi there, Googling through the archives it looks like I'm the ferst person to want this... My aim is to set up a voicemail application with a custom greeting before *AND AFTER* the punter has left the message. Right now the relevant section of my dialplan is like this: exten => 2,1,Playback(/media/asterisk/answerphone-en) exten => 2,n,VoiceMail(2000,s) exten =>

Hello, How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough variables in (within) my custom Asterisk application? I can't use chan_sip.c internal structures (such as sip_pvt) in my custom application, because there's no chan_sip.h and I can't include it into my application (maybe there's other way?). I can do like this: exten =>

I am building a new Asterisk server here at the office, and I'm wondering if there are any downsides to creating my dialplan with AEL. It seems more intuitive (to me), but I'm not sure if there are any pitfalls I need to be aware of first. We use this for internal extensions, 8 pots lines, and our answering service which gets about 500 incoming calls a day down our T1. Also, one more

Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460

hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell * ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.000000;txcount=83;rlp=0;rtt=14818.715000 * if any one know plese help me to or give any documentation link regards Dhaval -------------- next part

Hello all, I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP stats... Have you got any idea how to do it? Thanks I'm reading all G.107 ITU docs to retrieve something... I'm saving the SIP RTCP stats with: [macro-hangupcall] exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve

Hi All; I used A2Billing, basically it is nice and fine, but management possibilities is not that rich, so a lot of staff are need to be repeated that let the admin facing a problem of the needed time to do the task. Anyone advise for another open source prepaid billing that is rich by the management features? Also, I hope to find an open source Billing (prepaid and postpaid) that can work with

have problems with outgoing calls. When I tried this, the same way you did, I could make calles externally but had no audio each way reguardless of what I tried to pass to the sip provider. Best bet is to use what your sip provider can use or find another provider that that can do g722. That's what I did when I wanted to use g726. my2cents On Tue, Jun 29, 2010 at 2:42 PM, Mindaugas Kezys

Hi everyone Anyone please explain me How Account code is use for billing., Thanks Nikhil

Good morning everyone. I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? Thanks in advance, -- Rodrigo Lang Opening your mind - Just another Open Source site<http://openingyourmind.wordpress.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL:

CDR Stats Analyzer and Report generator It's a rework of famous Asterisk Stats written by Areski. The main goal for this project is to concentrate more on PDF reports (managers love them!). Later more functions will be added. Please test it and send suggestions how to improve it. Licence: GPL Examples, demo and more info on homepage: http://www.paskambink.lt/mcc Regards,

Are there any nice GUIs out there for Asterisk Realtime? Google doesn't yield much. I spent a day trying to get VoiceOne to work without much success. Thanks, Mike Clark

Good evening, Have you got any idea which prepaid application will be the best to do simple prepaid calls with a MySQL storage...? PS: I have a compiled by hand Asterisk 1.4.13 on a Debian Etch Thanks

managed without Realtime and I see no way how to put AEL into DB. Maybe it's possible? We are storing "exact-match" info into DB and all _X., etc stuff we have in extensions.conf. So no speed issues with large systems. Also: Any reason to "not" use extensions.conf? What AEL can do better then extensions.conf? Many people still use vi. Because it can do everything what

Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID calls, originating and transferring. A provider offers both SIP and IAX trunking. Cateris paribus, what is the preferred solution to choose? What points to consider? I can name the provider if this is not against this list policy--is it? Thanks, -kkm

Forgive me if this has been covered before. I did search but I was unable to find a reference. I am curious to know more about the possibility of using SQL to store voicemail as well as having more than one voicemail system accessing a central SQL database. Any information would be appreciated. Thank you all, in advance. -- Ein Bielaczyc <ebielaczyc at gmail.com> NOTICE: This E-mail

Hi, Is it possible to use Asterisk to record or monitor all conversation on standard PSTN PBX ? ASLAY