From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop working, failing withchan_sip.c:4083 retrans_pkt: Hanging up firstname.lastname@example.org:0 – no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions)..
I trying to solve one scenario:-As I can make call from mobile phone to my friend1. As he accept it, I put him on hold, & dial friend2. As he also accept it, I put him on hold & follow same procedure till friend6. The I click on Merge call & I can t..