Forward voicemail to email based on length [Asterisk Support] (3)
Struggling to make sense of sending DTMF and why DIAL is trying to make multiple calls? [Asterisk Support] (11)
Programming Languages and Libraries [Asterisk APIs] (3)
No SDP sent in outbound INVITE from Asterisk using PJSIP [Asterisk Support] (9)
Connecting an existing conference via PJSIP? [Asterisk Support] (4)
Tls sip clients cant register [Asterisk SIP] (14)
Saving CDR and CEL to postgres [Asterisk Support] (6)
Any example of using read application in Asterisk 15.5 with PHP AMI? [Asterisk Support] (8)
AGI-Script not responding over IAX2 channels [Asterisk Support] (1)
Connection from outside LAN denied [Asterisk SIP] (3)
Sending DTMF after a call is bridged [Asterisk Support] (3)
Call recording in Asterisk java not record all Calls ( 2 ) [Asterisk Support] (26)
Pjsip : registration status from the dialplan [Asterisk Support] (9)
Atxfer callback if busy [Asterisk Support] (2)
Unable to install Dahdi in centos 6.6 [Asterisk Support] (4)
Unable to install dahdi-linux-complete [Asterisk Hardware] (5)
Aasterisk ODBC simultaneously calls is mixing [Asterisk Support] (3)
Exceptionally long voice queue length queuing to Local [Asterisk Support] (7)
Is it possible to add any variable in cdr? [Asterisk Support] (5)
Chan_sip.c:10118 in process_sdp: Ignoring video stream offer because port number is zero (ONE WAY) [Asterisk SIP] (2)
Provider returns 403 after 300 seconds on qualify [Asterisk SIP] (3)
When I try to find the state of Endpoints it is showing unavailable [Asterisk SIP] (3)
How to count DTMF key enter by user by AMI? [Asterisk APIs] (3)
Asterisk 13.8.2 res_rtp_asterisk PJ ICE ERROR 370400 bad request ( 2 ) [Asterisk SIP] (21)
Sometimes extension not found even if it exists in dialplan and works just fine for most of time [Asterisk Support] (5)
Dahdi problem with asterisk 16 [Asterisk Hardware] (10)
Issue with calerid when using the SLA [Asterisk Support] (7)
Zoho error in Asterisk 15 - Dialed Channel=nul [Asterisk Integration] (3)
Port missing in R URI for incoming call [Asterisk SIP] (6)
Error loading codec_opus.so - GLIBC_2.14 required [Asterisk Support] (3)