Asterisk   Asterisk SIP


Topic Replies Activity
About the Asterisk SIP category 1 September 1, 2016
Endpoint ‘0000f30B0B02’: Could not create dialog to invalid URI ‘0000f30B0B02’ 7 December 13, 2019
[Solved] PJSIP Contact was deleted while dialing extension 3 December 13, 2019
Best practices for user registrations 10 December 13, 2019
Audio issues outside of the network 2 December 13, 2019
There was no sound on the call 6 December 13, 2019
No Audio - Not NAT related - ITSP to ASTERISK to ITSP (16.6.2 PJSIP) 5 December 12, 2019
Asterisk 16.4 every 20 some days chan sip wigs out 5 December 11, 2019
DTMF duplicate when RTP event have long delay between packets 2 December 10, 2019
PJSIP dial specific user-id at endpoint 1 December 10, 2019
Blind Transfer Drops after 30 second 4 December 9, 2019
Dialing to sip without creating a trunk - 2 December 8, 2019
Asterisk NAT Routing Libvirt (iptables) 3 December 7, 2019
Kamailio - Asterisk PJSIP trunk 1 December 5, 2019
Failed to handle incoming SDP. Session has been already disconnected 6 December 5, 2019
Asterisk Extconfig Multiple Sippeers 5 December 4, 2019
PJSIP Shutting transport 7 December 4, 2019
Choppy sound some time 2 December 4, 2019
SIP Transport: TCP (LAN:OK; Outside:OK) / TLS (LAN:OK; Outside: NOTOK) 2 December 2, 2019
Asterisk Tls configuration problems please help! 20 December 1, 2019
Make a call via registered account 9 November 29, 2019
Connect Asterisk 13 with Asterisk 1.8 10 November 28, 2019
Connect 2 Asterisk Servers SIP accounts 8 November 27, 2019
Read a custom parameter sent in registration request for later use 3 November 27, 2019
Sip trunk get unreachable and then reachable 5 November 27, 2019
Aasterisk 16.6.2 - Two channels? 5 November 25, 2019
Asterisk vmware esxi deployment RTP problems with one-way audio 17 November 22, 2019
Help in configuring Trunk using only SIP Signalling IP and WAN Ip 3 November 22, 2019
Register impossible after 1 day 6 November 21, 2019
Tls outbound proxy problem 1 November 20, 2019