Hi Jeff, On Thu, Jan 6, 2011 at 11:28 AM, Jeff LaCoursiere <[email protected]> wrote: > > > On Wed, 5 Jan 2011, James Lamanna wrote: > >> See the following SIP trace. >> Where in the world does Asterisk get port 1025 to respond to? >> This is asterisk 1.6.x. >> > > Hi James, > > I'm sure it would be the NAT translated port on the public side of the > customer's firewall...
Unfortunately its not. All clients are on symmetric NAT. Here's an ngrep trace, you can see the NAT port in the VIA is the same as the source port: U xxx.xxx.xxx.44:8155 -> xx.xxx.xxx.7:5060 NOTIFY sip:pbx1.warp2biz.com SIP/2.0..Via: SIP/2.0/UDP 192.168.1.127:8155;branch=z9hG4bK-4b50c77d..From: "zzzzzzzzzz" <sip:[email protected]>;tag=5281a88170274fa2o0..To: <sip:pbx1.example.com>..Call-ID: [email protected]: 14492 NOTIFY..Max-Forwards: 70..Con tact: "zzzzzzzzzz" <sip:[email protected]:8155>..Event: keep-alive..User-Agent: Cisco/SPA509G-7.4.6-0002fdff9097..Content-Length: 0.... # U xx.xxx.xxx.7:5060 -> xx.xxx.xxx.44:1025 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.127:8155;branch=z9hG4bK-4b50c77d;received=xx.xxx.xxx.44..From: "zzzzzzzzzz" <sip:[email protected]>;tag=5281a88170274fa2o0..To: <sip:pbx1.example.com>;tag=as62dac391..Call-ID: [email protected]: 14492 NOTIFY..User- Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Length: 0.... -- James > > j > >> Thanks. >> >> -- James >> >> >> <--- SIP read from zzz.zzz.zzz.44:9363 ---> >> NOTIFY sip:pbx1.mydomain.com SIP/2.0^M >> Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M >> From: "xxx-xxx-xxxx" >> <sip:[email protected]>;tag=467525dd6fac949do0^M >> To: <sip:pbx1.mydomain.com>^M >> Call-ID: [email protected]^m >> CSeq: 118907 NOTIFY^M >> Max-Forwards: 70^M >> Contact: "xxx-xxx-xxxx" <sip:[email protected]:9363>^M >> Event: keep-alive^M >> User-Agent: Cisco/SPA509G-7.4.6-0002fdff90a4^M >> Content-Length: 0^M >> ^M >> >> <-------------> >> [Jan 5 13:46:36] VERBOSE[3919] logger.c: --- (11 headers 0 lines) --- >> [Jan 5 13:46:36] VERBOSE[3919] logger.c: >> <--- Transmitting (no NAT) to zzz.zzz.zzz.44:1025 ---> >> SIP/2.0 200 OK^M >> Via: SIP/2.0/UDP >> 192.168.1.140:9363;branch=z9hG4bK-b9a860d3;received=zzz.zzz.zzz.44^M >> From: "xxx-xxx-xxxx" >> <sip:[email protected]>;tag=467525dd6fac949do0^M >> To: <sip:pbx1.mydomain.com>;tag=as0493c604^M >> Call-ID: [email protected]^m >> CSeq: 118907 NOTIFY^M >> User-Agent: Asterisk PBX^M >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO^M >> Supported: replaces^M >> Content-Length: 0^M >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > -- > _____________________________________________________________________ > -- 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 > -- _____________________________________________________________________ -- 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
