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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

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 =>

...n) == Everyone is busy/congested at this time (1:0/0/1) Because Realtime Engine is not able to find my Provider which is NOT DYNAMIC! No other settings changed. Same configuration files. res_config_mysql.so recompiled to 1.4.16.1. Please help or explain what's wrong! Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX

...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, Mindaugas Kezys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060402/9382d09b/attachment.htm

with newest Asterisk version.=20 When holidays will end more and more people will start to complain about = this. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -----Original Message----- From: asterisk-users-bounces at lists.digium.com = [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anthony = Messina Sent: Sunday, December 30, 2007 12:09 AM To: Asterisk Users Mailing Lis...

...angup $i " > /dev/null done # let asterisk to stop by itself sleep 5 # kill remainings killall -9 safe_asterisk killall -9 asterisk # start fresh and ready to work! /etc/init.d/asterisk start # clean rm -rf /tmp/f1 rm -rf /tmp/f2 Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090319/0fdf36ab/attachment.htm

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

...lles 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 <mkezys at gmail.com> wrote: > Try this: http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch > > Regards, > Mindaugas Kezys > > Kolmisoft UAB > VoIP Billing Solutions > e-mail: info at kolmisoft.com > URL: http://www.kolmisoft.com > > > -----Original M...

...tuff 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 they want. Same here with extensions.conf. Regards, Mindaugas Kezys http://www.kolmisoft.com > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of Steve Murphy > Sent: Tuesday, June 03, 2008 9:02 PM > To: Asterisk Users Mailing List - Non-Commercial Discus...

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 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

...gain. The call never goes to Agent2 (only when Agent1 is not connected/paused and similar) So basically right now when Agent1 does not take the call dialing is: Agent1 Agent1 Agent1..... I want: Agent1 Agent2 Agent3.... What could be the problem? Regards/Pagarbiai, Mindaugas Kezys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060621/f27b4226/attachment.htm

...application (maybe there's other way?). I can do like this: exten => _X,1,Set(PEERIP=${SIPCHANINFO(peerip)}) exten => _X,2,custom_app and read PEERIP with pbx_builtin_getvar_helper, but that's not an option for me. Any help? Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080325/57c600f2/attachment.htm

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

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

I lurk and comment a little on here and have been playing with * for a short while. I am interested in hearing about the pros and cons for using a database backend to Asterisk. My current setup is simple, out of the box with config files in /etc/asterisk and logs etc going into /var. I notice a great many of the contributors here seem to use a db backend (is this also called Real Time

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

...access caller's number with Asterisk CVS 1.0.12? In new version there are structure cid with field cid_num. And in 1.0.12 only callerid field which is equal to cid_name. I also tried to get it from chan->cdr->src but this is also the same as cid_name or callerid. Mindaugas Kezys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050104/71691735/attachment.htm

Hello to all Im looking for a billing tool for Asterisk, that works with PostgreSQL. All the tools I found in www.asteriskbilling.com just work with MySQL :( Do you know a nice billing tool for Asterisk with PostgreSQL? Thanks Joao Pereira

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