search for: mkezi

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 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 =>

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, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on

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, 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 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 =>

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, 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, 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, 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

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?

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, 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]: