Asterisk   Asterisk SIP


Topic Replies Activity
Problem with transfering call 2 January 20, 2020
PJSIP json UTF-8 error 6 January 19, 2020
Audio issues outside of the network 3 January 19, 2020
Best practices for user registrations 12 January 19, 2020
SIP over VPN - One Way Audio (Incoming only) 3 January 18, 2020
Detect sip busy 4 January 18, 2020
Symmetric RTP when one side only sends keep-alive RTP Packets 3 January 17, 2020
Is it possible to establish a SIP session without initial Audio 5 January 17, 2020
Asterisk as SIP client (register) over TCP 4 January 16, 2020
Endpoint ‘0000f30B0B02’: Could not create dialog to invalid URI ‘0000f30B0B02’ 9 January 15, 2020
Linksys SPA3000: Context switches to public context when CID is turned on 9 January 14, 2020
[Solved] PJSIP Contact was deleted while dialing extension 4 January 12, 2020
There was no sound on the call 7 January 12, 2020
No Audio - Not NAT related - ITSP to ASTERISK to ITSP (16.6.2 PJSIP) 6 January 11, 2020
DTMF duplicate when RTP event have long delay between packets 3 January 9, 2020
Blind Transfer Drops after 30 second 5 January 8, 2020
Dialing to sip without creating a trunk - 3 January 7, 2020
Asterisk NAT Routing Libvirt (iptables) 4 January 6, 2020
Failed to handle incoming SDP. Session has been already disconnected 7 January 4, 2020
Asterisk Extconfig Multiple Sippeers 6 January 3, 2020
PJSIP Shutting transport 8 January 3, 2020
Choppy sound some time 3 January 3, 2020
SIP Transport: TCP (LAN:OK; Outside:OK) / TLS (LAN:OK; Outside: NOTOK) 3 January 1, 2020
Asterisk Tls configuration problems please help! 21 December 31, 2019
Make a call via registered account 10 December 29, 2019
Connect Asterisk 13 with Asterisk 1.8 11 December 28, 2019
Connect 2 Asterisk Servers SIP accounts 9 December 27, 2019
Read a custom parameter sent in registration request for later use 4 December 27, 2019
Sip trunk get unreachable and then reachable 6 December 27, 2019
Aasterisk 16.6.2 - Two channels? 6 December 25, 2019