similar to: Background stress test

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

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

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

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

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

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, 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

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

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 =>

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, How can I go to the Leg B channel in Asterisk Dialplan _after_ call ends? I can use Dial G option to go to Leb B channel when call is answered, but how to go here when call ends? Is here any option/function in Dial Plan? Or should I use ast_bridged_channel(chan) to get bridged channel and try to retrieve data I need from internal structures using custom c module and Asterisk API?

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, We would like to share our webinar about Auto-Dialer Business Model (Telemarketing). It is educational video which we made for our clients and now we are sharing it with you. http://www.kolmisoft.com/how-to-start-a-VoIP-business/webinars/ Enjoy. NOTE: This is not attempt to sell you anything. No product or service is sold/marketed in the video. Regards, Kolmisoft Team

Hi, we bought some machines with 2 x quad core Xeon E5405 processors and installed centos 5.3 on them. My problem is that I can't get the cpuspeed service to work. No driver seems to claim responsibility for the throttling and the fallback "modprobe acpi_cpufreq" in the cpuspeed init script just yields a "No such device" message. According to the acpi information the

Hi all, I'm not sure if this is really rsync related so fell free to ignore this message ;) I have to server. The first one is rsync server et and the second one the client. The rsync server repo is around 325Go. Every time my client start rsync, both rsyncd on the server and rsync on the client entered "Uninterruptible sleep" state and I can't figure out why. Server : -

I am running the latest updated version of CentOS on a HP DL160 G6 server as a workstation with the Mate desktop, not Gnome. The only expansion card I have installed is a MSI Geforce GT710 graphics card driving two monitors. Unfortunately the computer locks up at random intervals: neither the mouse nor the keyboard work and I lose the SSH connection to the computer I have used to see if the

Hi, I am working with a westemere based server. I need to find out if the NIC installed there supports SR-IOV - what would be the best way to determine that? Regards, Kashyap hwconfig ------------- Summary: HP DL160 G6, 2 x Xeon E5620 2.40GHz, 15.7GB / 16GB 1333MHz, 1 x 500GB SATA System: HP ProLiant DL160 G6, C-2N/16/500, ySPEC 25.0 Processors: 2 x Xeon E5620 2.40GHz (HT

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 =>