Asterisk   Asterisk SIP

About the Asterisk SIP category (1)
Making sense of PJSIP, relationship between AORs and Endpoints (3)
Asterisk port/network issue (3)
Remote peers audio quality issue (2)
Calls are getting stuck with Pjsip and also some call-drop (7)
Problem with pjsip dial (5)
Transfer recalls via REFER NOTIFY events (3)
DNS trunk SIP with Alcatel OXE (3)
Wrong SDP details, RTP and NO Audio (14)
Asterisk 13 -> Linksys SPA 3000 -> PSTN (3)
Why my callerid display to the ip phone of my caller? (5)
SIP Trunk Connection issue (14)
Order of codecs in outgoing SDP (PJSIP) (9)
Pass SDP transparently (2)
Explicit outbound calls (no auth) (7)
Trying to connect devices in different networks (6)
Type=peer and still able to place calls (3)
Can I disable P-Asserted-Identity in dialplan dynamically to affect only current call? (2)
OpenSIPS Asterisk PJSIP Realtime Media Encryption (3)
In-Dialogue SIP OPTIONS (2)
Early Media Support (100rel/PRACK) on Asterisk 15.5 (7)
PJSIP2.4.5. unsuitable transport selected (7)
Behavior with both remotesecret and username (3)
PJSIP client_uri with @ in username (3)
TLS & RSTP for Telekom DeutschlandLAN sip trunk (9)
How to send differentiated MWI notifications to ISDN gateway? (2)
Asterisk Console logging "Request 'OPTIONS'..." each 1 minute (5)
Receiving no Calling number via SIP Trunk (17)
How to reload a specific SIP account (2)
Hold-Resume: Asterisk did not forward the call Hold reINVITE (7)