Asterisk Dialplan Using dialplan and have questions? This is the place for you! Asterisk WebRTC For things about WebRTC in Asterisk. Asterisk Fax This category is for discussion about faxing with Asterisk. Asterisk News News related to the Asterisk project. Asterisk Hardware Discussions about hardware used with Asterisk, such as telephony cards. Asterisk Distributions This category is for the discussion of distributions that use Asterisk underneath. Asterisk Embedded This category is for discussion related to Asterisk in embedded environments, such as running on ARM based devices. Asterisk Integration This category is for the discussion of integrating Asterisk in other platforms. Asterisk SIP Have a question about Asterisk's SIP functionality? Have a generic SIP question? This is the category for you! Asterisk Support If you are having difficulties installing, operating, upgrading or configuring Asterisk, post your issues in Support. Please make sure to check resources and to search the forum before posting. Most standard and basic questions have been asked before. Asterisk APIs Are you working with <a href="">AMI</a>, <a href="">AGI</a>, or <a href="">ARI</a>? Writing a custom application with Asterisk as the engine? Then this is the category for you! Asterisk Endpoints Have an endpoint that is connected to Asterisk and want to discuss something about it? This is the category for you! Asterisk General This category is a general catch-all for Asterisk questions that don't have a better categorization. If at all possible, try to see if your topic is better suited for one of the other categories before using this one. In particular, if you're having trouble with Asterisk, check out the <a href="">Asterisk Support</a> category.
About the Asterisk category [Asterisk] (1)
Passing dtmf in dial application [Asterisk Support] (4)
Cannot find kickstart [Asterisk Distributions] (2)
Issue when using playback application for outbound video call [Asterisk Support] (1)
FastAgi get_variable after GoSub [Asterisk APIs] (3)
Asterisk 11 -> 16 Upgrade results in 1 way voice [Asterisk SIP] (10)
Phpagi GET FULL VARIABLE [Asterisk APIs] (11)
Channel held open that Asterisk couldn't destroy [Asterisk SIP] (7)
Frequent Disconnection issue in Webrtc sipml5 [Asterisk WebRTC] (1)
TIMEOUT(absolute) is not working [Asterisk Support] (19)
ASTERISK bridge unintelligible calls and playback sounds [Asterisk Support] (2)
Hints are getting stuck (for BLF) [Asterisk SIP] (13)
TLS & RSTP for Telekom DeutschlandLAN sip trunk [Asterisk SIP] (8)
How to have a music in the same time of ringing during a call? ( 2 3 4 ) [Asterisk Support] (64)
Call file retry if no valid answer on ivr [Asterisk Support] (6)
ARI connection problem [Asterisk APIs] (4)
Concurrent Call limit on static agents [Asterisk Support] (5)
Bad audio quality, 3KHz noise [Asterisk Hardware] (4)
Asterisk Originate Command With clid [Asterisk APIs] (7)
Accessing a variable from one context to another context in dialplan [Asterisk Dialplan] (5)
Real Time in Asterisk 16 [Asterisk Support] (3)
Reading the combination of the letters and characters in asterisk gateway interface [Asterisk APIs] (5)
AMI Pause Mixmonitor - Not Monitor [Asterisk Support] (2)
Unwanted pause/delay between dialplan elements [Asterisk Support] (4)
Pjsip cipher 256 [Asterisk SIP] (19)
Moving Asterisk from Old Server Hardware to New Server Hardware [Asterisk Support] (4)
Passing the value of a variable from dialplan to the agi file [Asterisk APIs] (3)
During an active session caller can offered to change pin / passcode in database [Asterisk APIs] (5)
AMI PHP Data collector [Asterisk APIs] (2)
"WARNING T.30 Page did not end cleanly"? [Asterisk Fax] (1)