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    Since yesterday I have a stuck channel on my Asterisk server and I do not know how to eliminate it: Message/ast_msg_queu macro-dial-one       s                  59 Up      Dial         PJSIP/1218/sip:1218@19..

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all,No audio flows after holdIm getting this below error after unholding the call.chan_sip.c:10088 process_sdp: Declining non-primary audio stream:audio 14670 UDP/TLS/RTP/SAVPF 107 103 104 9 0 8 106 105 13 110 112 113 101is it a codec issue or someth..

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