Issue while unhold a call. asterisk13.23 + SIPML5

Asterisk Asterisk SIP
1

Hi Team,

I am facing a issue while hold/resume a call using sipml5 phone.
while unholding a call I got below mentioned error on my asterisk cli. and customer end is unable to hear anything. please help me.

chan_sip.c:10393 process_sdp: Declining non-primary audio stream: audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126

0x7f97a8080000 – Strict RTP learning after remote address set to: 192.168.1.6:58811
– Stopped music on hold on DAHDI/i1/0XXXXXXXXXX-71

Thanks in Advance

2

Please provide “sip set debug on” output contain at least the SDP exchanged in both directions for the initial INVITE, the re-INVITE to held, and the failing re-INVITE.

3

It seems as though the WebRTC client isn’t actually unholding, but is instead adding a second audio stream which is unsupported. The “sip set debug on” mentioned by David will confirm this.

4

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.