Due to circumstances beyond my control, I'm stuck getting a SIP phone working outside the office network. The issue I'm experiencing at the moment is when I call out from the remote phone to say, my cellphone, my cellphone rings but on the remote phone there is no audio (not even a ringing.. tone). When I answer, I get no audio in either direction. About 5 seconds later, the call terminates automatically on my cell. On the remote phone, call stays established until hung up normally.
The asterisk log (shown below, [pastebin link of whole log]) shows the NAT'ed Internal LAN ip of the phone -- 10.2.10.113, not the WAN IP. Should it be showing the WAN IP? Is this correct behaviour?
Is my diagnosis that this is a NAT problem correct? If yes, how can I resolve this? Unfortunately due to the location of the devices getting a wireshark capture is proving to be a little difficult.
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/DVX/<CELL NO>
-- SIP/DVX-00000aa9 is making progress passing it to SIP/260-00000aa8
-- SIP/DVX-00000aa9 answered SIP/260-00000aa8
[2015-06-08 10:26:27] WARNING[2270]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission 1019324208@10.2.10.113 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2015-06-08 10:26:27] WARNING[2270]: chan_sip.c:4198 retrans_pkt: Hanging up call 1019324208@10.2.10.113 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/260-00000aa8", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/260-00000aa8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/260-00000aa8", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/260-00000aa8", "") in new stack
[1]: http://pastebin.com/tKLRux6Z
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