![]() ![]() VERBOSE res_rtp_asterisk.c: 0x7f05cc0860d0 - Strict RTP switching to RTP target address 212.117.203.158:32406 as source VERBOSE res_rtp_asterisk.c: 0x7f05cc0773a0 - Strict RTP learning complete - Locking on source address 91.67.195.16:58920 VERBOSE res_rtp_asterisk.c: 0x7f05cc0773a0 - Strict RTP qualifying stream type: audio VERBOSE bridge_channel.c: Channel PJSIP/hativ-00000002 joined 'simple_bridge' basic-bridge VERBOSE bridge_channel.c: Channel PJSIP/hativ-voip-00000003 joined 'simple_bridge' basic-bridge Log (see the delay between seconds 11 to 13) VERBOSE app_dial.c: PJSIP/hativ-voip-00000003 answered PJSIP/hativ-00000002 The Asterisk is in a data center, the browser / client is behind NAT. But everything is fine with incoming calls. Unfortunately, I often don't hear the first few seconds when I call someone. I run an Asterisk 16 installation and a WebPhone based on SIP.js. ![]()
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