Asterisk Dialplan Using dialplan and have questions? This is the place for you! 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 WebRTC For things about WebRTC in Asterisk. Asterisk Hardware Discussions about hardware used with Asterisk, such as telephony cards. Asterisk Embedded This category is for discussion related to Asterisk in embedded environments, such as running on ARM based devices. Asterisk News News related to the Asterisk project. Asterisk Fax This category is for discussion about faxing with Asterisk. Asterisk Integration This category is for the discussion of integrating Asterisk in other platforms. Asterisk Distributions This category is for the discussion of distributions that use Asterisk underneath. 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. Asterisk Endpoints Have an endpoint that is connected to Asterisk and want to discuss something about it? This is the category for you!
About the Asterisk category [Asterisk] (1)
Help Wanted! For Hire! [Asterisk Support] (5)
Avaya 4602SW+ web interface [Asterisk Endpoints] (1)
PJSIP No Matching endpoint found + Failed to authenticate (inbound) [Asterisk SIP] (3)
I need to configure a SIP Trunk with different SIP Signaling IP and SIP Media IP [Asterisk SIP] (19)
Asterisk 13 PJSIP sometime sounds works some time not [Asterisk SIP] (1)
Best Strategy - Multiple Company Trunks and Phone Numbers [Asterisk General] (8)
Incoming caller is muted after being placed on hold [Asterisk Support] (3)
Ring Groups & CDR Entries [Asterisk Support] (3)
Error to connect a PBX central "Intelbras Impacta" in Asterisk PJSIP ( 2 ) [Asterisk SIP] (25)
No outbound calls with new install [Asterisk Distributions] (2)
Retransmission Timeout Reached for Critical Packet, Configurable? [Asterisk SIP] (2)
[SOLVED] Error when I try to enable the TLS transport in PJSIP Asterisk [Asterisk SIP] (3)
During an active session caller can offered to change pin / passcode in database [Asterisk APIs] (6)
${SIPDOMAIN} and REFER [Asterisk SIP] (1)
Asterisk -> Spa 3000 -> PSTN LINE [Asterisk SIP] (7)
Registration fail between two asterisk server using PSJIP [Asterisk SIP] (5)
Multiple Sip Trunks inbound not working [Asterisk SIP] (5)
Asterisk SFU Video Conferencing [Asterisk WebRTC] (10)
Starting an Application during a call [Asterisk APIs] (19)
Add a=rtpmap and a=fmtp [Asterisk SIP] (10)
Asteriskk 15 spontaneous restsart [Asterisk Support] (2)
Logging and the console [Asterisk Support] (1)
Asterisk suddenly fails to register SIP [Asterisk SIP] (15)
How to use lock/unlock in dialplan asterisk 13? [Asterisk Dialplan] (4)
[SOLVED] Unable to Provision Digium phone with Asterisk 16 using DPMA/PJSIP [Asterisk Endpoints] (3)
One way audio with Asterisk 15 and a WebRTC client [Asterisk WebRTC] (5)
Asterisk Integration with Apptivo CRM [Asterisk Integration] (7)
Unable to do SIP calling between WebRTC clients ( 2 3 4 5 ) [Asterisk Support] (98)
Passing Variables Through Dialplan [Asterisk Dialplan] (6)