Refer => re-invite [Asterisk General] (4)
Incoming fax problem fax_spandsp & asterisk 13 [Asterisk Fax] (2)
Cannot match incoming calls to peers (peers exists, ip addresses match) [Asterisk SIP] (10)
Channel count in struct ast_endpoint continuously increasing and causing memory leak in case of bridge application [Asterisk Endpoints] (16)
Queue variables [Asterisk Dialplan] (19)
DTMF troubleshooting [Asterisk Support] (10)
Dtmf sequence in ARI [Asterisk APIs] (4)
Connecting Two Asterisk System [Asterisk Support] (4)
P-Early-Media header (RFC 5009) [Asterisk SIP] (2)
Extension that forwarded a call on the queue not receiving hangup at the end of forward [Asterisk APIs] (1)
Asterisk realtime configuration with mysql and odbc [Asterisk Support] (6)
Linx version question [Asterisk Support] (2)
Asterisk not playing unavailable message from odbc [Asterisk Support] (3)
Astrix training course [Asterisk Support] (3)
SIP INFO is not present in Allow header [Asterisk SIP] (4)
Google sip trunk [Asterisk SIP] (4)
Asterisk sending CDR to legacy CDR/SMDR collector [Asterisk Integration] (1)
PHPAGI Play audio without answering [Asterisk Support] (7)
Calls hangup 40 -50 Second via API or Direct [Asterisk Support] (2)
Asterisk 13.13 crash/restart bug ?! (CLOSED) [Asterisk Support] (4)
ARI error on hangup [Asterisk APIs] (1)
How to shutdown span via CLI or API? [Asterisk APIs] (1)
Sip and pjsip configuration issue? [Asterisk Support] (10)
Support for RFC5168 in Asterisk13 [Asterisk Support] (4)
Making sense of PJSIP, relationship between AORs and Endpoints [Asterisk SIP] (3)
Cdr show in cli? [Asterisk Support] (8)
Digium TE420 Quad Span Card only uses one processor core [Asterisk Hardware] (2)
Asterisk Ports & Phones [Asterisk Hardware] (4)
Dialing between web browser and soft phone [Asterisk WebRTC] (13)
Dial # before entering extension [Asterisk Dialplan] (10)