When remote party disconnects the call, BYE goes to
PUBLIC-IP port 5060 and router blocks de request. I assume
vendor sends BYE to 5060 because it is a new transaction
-----------------------------------------------2--------------------------------------------------------------------------------------------------
If I use this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 it
work !!!!!!
When I dial a call, INVITE / ACK / Trying / OK goes fine
because they are part of the same transaction
When remote party disconnects the call, BYE goes to
PUBLIC-IP port 52548 and router forward the request to
Kamailio. Since there is an open connection.
I need to find the way to find the way to advertise the
public port internet router is doing NAT (PAT).
---------------------------------------------------------------------------------------------------------------------------------------------------
This trace is a call that worked fine because I
included line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
This trace is an INVITE with this line:
listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060 ->
VENDOR-IP:5060
INVITE
sip:NUM-DESTINATION@VENDOR-IP SIP/2.0
Record-Route:
<
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB
A-;nat=yes>
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
Max-Forwards: 69
From: NUM-SOURCE
<sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To:
<sip:NUM-DESTINATION@sip.VENDOR-IP>
Contact:
<sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1>
Call-ID:
329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
User-Agent: Kamailio
Date: Wed, 13 Jan 2016 19:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255
Trying.....
2016/01/13 20:10:15.842055 VENDOR-IP:5060 ->
PRIVATE-IP-KAMAILIO:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548
Via: SIP/2.0/UDP
PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
From: NUM-SOURCE
<sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
To:
<sip:NUM-DESTINATION@VENDOR-IP>
Call-ID:
329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 102 INVITE
Server: kamailio
Content-Length: 0
And finally a BYE
2016/01/13 20:10:28.545526 VENDOR-IP:5060 ->
PRIVATE-IP-KAMAILIO:5060
BYE
sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1
SIP/2.0
Via: SIP/2.0/UDP
VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0
From:
<sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93
To: "NUM-SOURCE" <
sip:NUM-SOURCE@norvoz.es>;tag=as3b72a453
Call-ID:
329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
CSeq: 28731 BYE
Max-Forwards: 69
Route:
<
sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na
yes>
Reason: Q.850;cause=16
Content-Length: 0
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
-----------------------------------------------------------------------------------------------------------------------------------------------------------------------
Finally, It is finally working because I hardcoded
NATīd port.
I would like to find a way to avoid setting the port in
"hard".
Thank you