[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue

[Scott Griepentrog]

It would appear that you have the Asterisk server on a public IP address,
your two endpoints are behind a NAT, and you have rewrite_contact enabled
in pjsip.conf.

In which case, what you are seeing is correct.  In order to be able to send
a call to an extension where it is behind NAT, Asterisk must update the
contact to have the current IP and port that the phone registered via (i.e.
the WAN IP of the NAT, and the WAN port that it is retaining state for).

On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:

> I am following the instructions in
> https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality 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
> (caller), but Bob's phone (callee) 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, but is shown here for clarity:
>
> root at asterisk13FFP:/var/log/asterisk# more /etc/asterisk/extensions.conf
> [from-internal]
> exten=>6001,1,Dial(PJSIP/demo-alice)
> exten=>6002,1,Dial(PJSIP/demo-bob)
> exten=>6003,1,Answer()
> same =>6003,n,Playback(hello-world)
> same =>6003,n,Hangup()
>
>
> 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 (both) show IP address
> 146.115.163.234. Any help is deeply appreciated. Thanks.
>
> asterisk13FFP*CLI> pjsip show endpoints
>
>  Endpoint:  <Endpoint/CID.....................................>
>  <State.....>  <Channels.>
>     I/OAuth:
>  <AuthId/UserName...........................................................>
>         Aor:  <Aor............................................>
>  <MaxContact>
>       Contact:  <Aor/ContactUri...............................>
>  <Status....>  <RTT(ms)..>
>   Transport:  <TransportId........>  <Type>  <cos>  <tos>
>  <BindAddress..................>
>    Identify:
>  <Identify/Endpoint.........................................................>
>         Match:  <ip/cidr.........................>
>     Channel:  <ChannelId......................................>
>  <State.....>  <Time(sec)>
>         Exten: <DialedExten...........>  CLCID: <ConnectedLineCID.......>
>
>  =========================================================================================
>
>  Endpoint:  demo-alice
> Unavailable   0 of inf
>      InAuth:  demo-alice/demo-alice
>         Aor:  demo-alice                                         1
>       Contact:  demo-alice/sip:demo-alice@*146.115.163.234*:38519
>  Unknown               nan
>
>  Endpoint:  demo-bob                                             Not in
> use    0 of inf
>      InAuth:  demo-bob/demo-bob
>         Aor:  demo-bob                                           1
>       Contact:  demo-bob/sip:demo-bob@*146.115.163.234*:38321;tra
>  Unknown               nan
>
>
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Scott Griepentrog
Digium, Inc · Software Developer
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