I am following the instructions in wiki.asterisk.org/wiki/display/ … ctionality and I am trying to make a call from extension Alice (6001) to extension for Bob (6002). When I make the call, I can hear the ringing on Alice’s phone, but Bob’s phone doesn’t ring, or show a call coming in from Alice. My setup and environment is as follows: Alice, Bob and Asterisk all in the same 192.168.1.0/24 network, and they are able to register to the Asterisk server running 13.1.0/PJSIP. The rest of the configuration is the same as the aforementioned wiki page.
What I do observe is that I when I request the output of pjsip show endpoints, I get Contact information for the two SIP peers that have registered different from their actual IP addresses. I suspect this has something to do with their calls being routed elsewhere. If my assumption is correct–how do I fix this? Alice should be at 192.168.1.50 and Bob should be at 192.168.1.149, instead, they show IP address 184.108.40.206. Any help is deeply appreciated. Thanks.
[code]asterisk13FFP*CLI> pjsip show endpoints
Endpoint: <Endpoint/CID…> <State…> <Channels.>
Contact: <Aor/ContactUri…> <Status…> <RTT(ms)…>
Transport: <TransportId…> <BindAddress…>
Channel: <ChannelId…> <State…> <Time(sec)>
Exten: <DialedExten…> CLCID: <ConnectedLineCID…>
Endpoint: demo-alice Unavailable 0 of inf
Aor: demo-alice 1
Contact: demo-alice/sip:firstname.lastname@example.org:38519 Unknown nan
Endpoint: demo-bob Not in use 0 of inf
Aor: demo-bob 1
Contact: demo-bob/sip:email@example.com:38321;tra Unknown nan