I am upgrading an Asterisk 11 server to Asterisk 16. Mainly just moved over configuration files and everything works…ALMOST. I get 1 way voice through the new Asterisk 16 server. Destination to Origin voice works fine, but Originating caller to destination does not work. I even get this error [2018-11-15 21:00:28] WARNING: chan_sip.c:4114 retrans_pkt: Retransmission timeout reached on transmission REPQJ3BH7JBEXG2MEYNM7CD4XU@ for seqno 102 (Critical Response).
There is no NAT on the originating end or the terminating side servers. Insecure is still set to invite,port and nat is set to No. These setting all worked with Asterisk 11. The asterisk 11 server can still route a call from the same originator to the same destination. The *16 server routes the call, and everything works fine except for the 1 way voice.
Can anyone think of a change from *11 to *16 that could cause this?
Thank you for the response. This is an AWS server so an important feature is externip=
AND it is important to note that the parameter name is externip NOT externalip which slipped me up for a couple of hours of troubleshooting. Working now.
That it is wrong on chan_sip.so media address and signaling address are unique and need to eb defined on the general section, if you want to use different values per sip devices you need to use pjsip channel driver
Use the command sip show settings to view the detailed list of the configuration of the SIP channel. and review the section called Network setting, verify if you have externip listed as parameter or check your sample configuration files, but if you say it works just keep using it