similar to: How to get LEG B channel info?

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

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

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

As Asterisk has inner problems and channels very often locks we have such script to restart Asterisk each midnight. We (our clients) mostly use v1.4.18.1. We can't upgrade to newer versions because there are too much changes which would brake our system (realtime/sip/iax2/cdr/etc/etc). Script soft hangups all alive channels in dirty way then kills Asterisk and starts it up. Hope

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

suite our billing needs. That was on 1.4.xx, we are not using 1.6+ Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: info at kolmisoft.com URL: http://www.kolmisoft.com Find us on Facebook -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nikhil Sent: Monday, November 22, 2010 7:20

Hello, We did small test with sipp to test Asterisk Background command capability. Our goal was 700 sim. calls on HP Proliant DL160 G5 E5405 1 x Quad Core Xeon 2Ghz 2 Gb RAM Asterisk 1.4.18.1 Centos 5.2 We reached more then 1000 when our network (100mbps) become a bottleneck. As we achieved our goal - no further testing was performed. As conclusion - we are very

Hello, We are glad to announce new release of our advanced billing and routing package for Asterisk - MOR v0.7 It is complete solution for VoIP billing and routing for advanced and start-up telecoms, carriers, voip calling card operators and ISPs. Demo available online, as LiveCD or as InstallCD. Contact us for more details. More info: http://www.kolmisoft.com What is new in

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 =>

And after reload ALL your phones are unreachable for 2 minutes! Imagine you have several thousands devices unreachable for 2 minutes. How much calls will fail during that time? Regards, Mindaugas Kezys Kolmisoft UAB=20 VoIP Billing Solutions e-mail: info at kolmisoft.com URL: http://www.kolmisoft.com -----Original Message----- From: asterisk-users-bounces at lists.digium.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]:

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

1. http://bugs.digium.com/view.php?id=12362 2. http://bugs.digium.com/view.php?id=12925 3. http://bugs.digium.com/view.php?id=12921 Also how do you go about changing details for device in DB and not using "sip realtime prune PEER" + 'sip reload'? Without that your changes to devices are not active. Good luck! Regards, Mindaugas Kezys http://www.kolmisoft.com >

Hello, We use Asterisk Realtime for our billing software. 200+ installations of Asterisk with Realtime, but I see this for the first time. Asterisk 1.4.17, Addons 1.4.5, No patches, no NAT - just plain simple installation. With debug I can see: [Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:662 mysql_reconnect: MySQL RealTime: Everything is fine. [Jan 30 22:38:21] DEBUG[27885]:

Hello, I got situation which is unclear for me, hope somebody could explain this. A calls to B INVITE sent from A to B B responds with 100 Trying B responds with 183 Progress After 10 seconds: Asterisk CLI: Scheduling destruction of SIP dialog '..' in 32000 ms (Method: INVITE) Asterisk sends CANCEL _instantly_ B responds with 200 OK and 487 Request Terminated Asterisk confirms 102 ACK

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 =>

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,

Hello, I'm trying to setup a queue where call goes from agent to agent in strictly set order. I have queue (roundrobin): Agent1 penalty 1 Agent2 penalty 2 Agent3 penalty 3 When I call to this queue Agent1 rings. If this agent does not take the call, after set timeout same Agent1 is dialed again. The call never goes to Agent2 (only when Agent1

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

Hello, How can I 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