two observations the extensions are all behind a NAT.
analyzing the packet capture at the time of the error connecting from extension to extension in assisted transfer, NAT is not being done in only one of the directions
In the PJSIP Profile, try disabling or enabling the “Direct Media” parameter. Additionally, disable the option “Disable NAT Direct Media.”
Remember to define the NAT settings in the PJSIP Settings module, and in the RTP Settings module, try disabling all the checkboxes but the “ICE Support” checkbox because it is required for WebRTC.
I’ll even take advantage of it, and then in a test environment enable pjsip and simulate this same environment to see if there’s the same problem and I report it here.
SIP protocol after updates to USE Update I’m with the following behavior that I couldn’t solve.
External call ----> extension answers ----> ok
External call ----> extension answers ----> assisted transfer “*2” → ok
External call —> queue → outgoing calls normally, but when directing to an agent the call is ended.
when checking the logs I found the following.
> 0x7f0c400ee700 -- Strict RTP learning complete - Locking on source address 200.201.197.137:7908
[2021-11-06 12:41:38] WARNING[2168]: chan_sip.c:4143 retrans_pkt: Retransmission timeout reached on transmission uEk0B4b7cJ2c8114b17c3c35101394b39830c2e980@200.201.197.137 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2021-11-06 12:41:38] WARNING[2168]: chan_sip.c:4167 retrans_pkt: Hanging up call uEk0B4b7cJ2c8114b17c3c35101394b39830c2e980@200.201.197.137 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
-- Stopped music on hold on SIP/imp27713995-000000de
the ip address that appears in the log is the ip address of the extension that is ringing and when the call is answered, it is terminated.
when I return the parameter to “Yes, If no Nat”
External call ----> extension answers ----> ok
External call ----> extension answers ----> assisted transfer “*2” → No audio call between extensions → after transferring external call the audio returns
External call —> queue → ok.
already according to the guidelines valid the same tests with the PJsip channel
extension to extension connections ----> ok (note the two extensions are in different networks)
external call —> direct to extension -->> ok
external call —>>> queue answers —> transfer to agent —>> audio only one way.