Type=peer and still able to place calls
Can I disable P-Asserted-Identity in dialplan dynamically to affect only current call?
OpenSIPS Asterisk PJSIP Realtime Media Encryption
In-Dialogue SIP OPTIONS
Early Media Support (100rel/PRACK) on Asterisk 15.5
PJSIP2.4.5. unsuitable transport selected
Behavior with both remotesecret and username
PJSIP client_uri with @ in username
TLS & RSTP for Telekom DeutschlandLAN sip trunk
How to send differentiated MWI notifications to ISDN gateway?
Asterisk Console logging "Request 'OPTIONS'..." each 1 minute
Receiving no Calling number via SIP Trunk
How to reload a specific SIP account
Hold-Resume: Asterisk did not forward the call Hold reINVITE
Channel held open that Asterisk couldn't destroy
PJSIP/Dial Congestion timeout
Asterisk behind Firewall at home , try to call from my mobile
Use third party sip
Tcptls.c: FILE * open failed!
Pjsip cipher 256
Client side TLS certificates
PJSIP_HEADER add - header not sent
Forwarding register to another server
Intermittent one way voice with Asterisk 13, 1.4 work no issue
How to log PJSIP Endpoints
PJSIP issue with TEL (RFC 3966)
Received Response "Forbidden"
Asterisk 11 -> 16 Upgrade results in 1 way voice
Hints are getting stuck (for BLF)
Sending Q reason on dial timeout
← previous page
next page →