search for: nat5

...mode : Unknown T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : Yes TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No Outb. proxy : nat5.babytel.ca DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : sip.babytel.ca Addr->IP : 198.38.7.11:5060 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 1234567890 SIP Options : (none) Codecs : 0...

...; DirectMedia : No > PromiscRedir : No > User=Phone : No > Video Support: No > Text Support : No > Ign SDP ver : No > Trust RPID : No > Send RPID : Yes > TrustIDOutbnd: Legacy > Subscriptions: Yes > Overlap dial : No > Outb. proxy : nat5.babytel.ca > DTMFmode : rfc2833 > Timer T1 : 500 > Timer B : 32000 > ToHost : sip.babytel.ca > Addr->IP : 198.38.7.11:5060 > Defaddr->IP : (null) > Prim.Transp. : UDP > Allowed.Trsp : UDP > Def. Username: 1234567890 > S...

I'm trying to configure SIP trunking. Now, I'm referencing "Asterisk the definitive guide", 4th ed. While I don't have the page handy, I was reading the suggestion to try SIP to SIP before proceeding to outside connectivity. I'm aware that SIP trunking is a construct, but am, obviously, learning the system. What I'd like to do is from the CLI "ping"

...Contact: <sip:16046289850 at 96.48.217.39:5060>;expires=55 Content-Length: 0 <-------------> [Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29] --- (8 headers 0 lines) --- [Mar 23 19:25:29] NOTICE[4067] chan_sip.c: Outbound Registration: Expiry for nat5.babytel.ca is 55 sec (Scheduling reregistration in 40 s) [Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29] Really destroying SIP dialog '5e070a0021f200c72308ddad6fe2521c at 192.168.0.99' Method: REGISTER [Mar 23 19:25:44] VERBOSE[4003] asterisk.c: [Mar 23 19:25:4...