Asterisk   Asterisk SIP


Can't make internal calls (14)
Asterisk Bandwidth.com configuration for incoming call (4)
Sip peer status (1)
What happens if name in externip = hostname is not resolvable at moment of call (1)
Implementing PJSIP Device State Exchange (6)
SRTP unprotect failed on SSRC 802983612 because of authentication failure 160 (4)
How to configure Asterisk to support emergency call for more than 20 seconds? (2)
Playback() no sound (5)
Asterisk Codec negotiation problem (4)
Pjsip trunk receive 401 after time (4)
Asterisk Register on Invite (3)
PJSIP transport selection logic (14)
Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) (5)
Asterisk inbound configuration with gsm (2)
Erroneous Invite with "%40" character (3)
How to send custom data form asterisk to sipml5 (6)
Caller ID pjsip (1)
Asterisk 15.x changing Record-Route set by Proxy (7)
Box with 2 interfaces, wrong one chosen in Contact header (6)
Communicate between Sipp UAC and UAS via Asterisk (2)
IP authentication with provider whose Invites originate from large IP range (7)
C function which sets contact header in '200 OK' Message Header ( 2 ) (30)
Oneway audio and reinvite (6)
No audio asterisk behind pfsebse (4)
Starting to use PJSiP to test Asterisk (5)
Error when my asterisk is configured in "offline" mode (1)
CHANUNAVAIL issue at outgoing (3)
Contact:Public-IP VS Contact:Private-IP in '200 OK' Asterisk response to an INVITE (1)
Config.c:1942 process_text_line: parse error: No category context for line 1 of /etc/asterisk/sip.conf (3)
Asterisk sip fromdomain add hostname (5)