Archives : July-2015
I have asterisk installed on CentOS with phpagi. Also I have PRI card connect to it. its possible to show the sip number when calling from sip number to external number thru the PRI, instead of showing the PRI number show the SIP number..
When configuring an extension on Asterisk we use the Syntax insecure=very or insecure=port etc. I did some research on Internet and I found that this is used to authenticate the peers, based on their IP/port. But I couldnt understand whats the differe..
I have a huge problem with a 1.8.11.0 Asterisk instance not logging CEL events with the correct eventtimes.Im logging via ODBC to MariaDB 15.1 Distrib 10.0.20-MariaDBIm logging into a MyISAM table.If I start the Asterisk instance, logged times are corre..
I would like to encrypt password between Asterisk servers and clients. is there an easy way to do so? I am running Asterisk 1.8.22.0 built on CentOS 6.3Tha..
All, Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Hereis my sip.conf[general]context = demo;Default context for incoming callsbindport = 5060;UDP Port to bind to (SIP standard port is 5060)bindaddr = 0.0.0.0..
Im planning on upgrading to Asterisk 13.4 soon and am looking for the corresponding Siren7 codec.Where do ..
I am running Asterisk 11 on CentOS 6.x. I have configured several queues as follows in extensions.conf:exten => s,1,Queue(myqueue,rtnC,18)same => n,Background(user_unavail)same => n,WaitExten(10)exten => 1,1,Voicemail(1111@my-vm,s)This rings the pho..
I dont know if this is something asterisk can do at the moment but on my setup, it does not.What I intend to do is, while a client is in a call, it will send an in-dialog re-invite to asterisk (after changes on the client i.e. IPaddress). Asterisk sho..
Any particular reason CentOS 7 repos arent available?Im finding integration issues with CentOS 6s ancient versions of MySQL and PHP with third party app..
Im following this tutorial (https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5) to deploy WebRTC support but Im having an issue with RTP when the WebRTCsoftphone is behind NAT.In my scenario, the Asterisk server is running a pub..