Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is now
[transport-udp] type=transport protocol=udp bind=0.0.0.0 local_net=192.168.1.0/24 ;Templates for the necessary config sections [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=ulaw [auth_userpass](!) type=auth auth_type=userpass [aor_dynamic](!) type=aor max_contacts=1 remove_existing=yes ;Definitions for our phones, using the templates above [demo-alice](endpoint_internal) auth=demo-alice aors=demo-alice mailboxes=box_a rewrite_contact=yes [demo-alice](auth_userpass) password=demo-alice ; put a strong, unique password here instead username=demo-alice [demo-alice](aor_dynamic) [demo-bob](endpoint_internal) auth=demo-bob aors=demo-bob mailboxes=box_b rewrite_contact=yes [demo-bob](auth_userpass) password=demo-bob ; put a strong, unique password here instead username=demo-bob [demo-bob](aor_dynamic) Thank you for your help! On Thu, Jan 8, 2015 at 11:48 AM, Scott Griepentrog <[email protected]> wrote: > 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 < > [email protected]> 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@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 >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > [image: Digium logo] > Scott Griepentrog > Digium, Inc · Software Developer > 445 Jan Davis Drive NW · Huntsville, AL 35806 · US > direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 > Check us out at: http://digium.com · http://asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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