Hi Jeff,
On Thu, Jan 6, 2011 at 11:28 AM, Jeff LaCoursiere j...@sunfone.com 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: zz
sip:zzz...@pbx1.example.com;tag=5281a88170274fa2o0..To:
sip:pbx1.example.com..Call-ID: c914b8d-532f2...@192.168.1.127..cseq:
14492 NOTIFY..Max-Forwards: 70..Con
tact: zz sip:zzz...@192.168.1.127:8155..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:
zz sip:zzz...@pbx1.example.com;tag=5281a88170274fa2o0..To:
sip:pbx1.example.com;tag=as62dac391..Call-ID:
c914b8d-532f2...@192.168.1.127..cseq: 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-
sip:xxx...@pbx1.mydomain.com;tag=467525dd6fac949do0^M
To: sip:pbx1.mydomain.com^M
Call-ID: 707176dd-38f47...@192.168.1.140^m
CSeq: 118907 NOTIFY^M
Max-Forwards: 70^M
Contact: xxx-xxx- sip:xx...@192.168.1.140:9363^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-
sip:xx...@pbx1.mydomain.com;tag=467525dd6fac949do0^M
To: sip:pbx1.mydomain.com;tag=as0493c604^M
Call-ID: 707176dd-38f47...@192.168.1.140^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
--
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asterisk-users mailing list
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--
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-- 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:
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