sstasterisk
Member
- Joined
- Jul 12, 2019
- Messages
- 82
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Exten to Exten dial works, but I receive no audio on both ends of the call
Thanks; Giving opensips a try totally new for me, how would I do sip capture?Cool story bro
You should troubleshoot that by looking at the opensips logs as well as SIP captures.
Right this moment traveling, no reliable internet connection, but here is my /etc/default/rtpproxy https://pastebin.com/ybc20Q8DAuth required is a normal part of the handshake; it is the proxy telling your phone that a password is required to make calls. Your phone replies with the username/password and the call proceeds. In the sngrep, take a look at the INVITE that is going from the proxy to the called phone. What IP address is being offered in the SDP portion of the INVITE (the lower portion; the IP address is on a line that starts with c= ) ? That is the IP address that the called phone will send the audio to. If it's the IP of the calling phone (that is, you are not using rtpproxy/rtpengine on your proxy to relay traffic), you need to make sure the phones can communicate directly with each other without being blocked by a firewall or NAT. If it's the IP of the proxy, you need to make sure rtpproxy is running so that your audio traffic is relayed.
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