Hi Alex
1.- Kamailio -> 172.26.101.50:8002 (Floating IP)
Asterisk -> 172.26.101.10:5080
2.- Transmitting (no NAT) to 192.168.0.170:8002:
ACK sip:110@IP_PUBLIC_IP:5066 SIP/2.0
Via: SIP/2.0/UDP 172.26.101.10:5080;branch=z9hG4bK0a330ae6
Route: sip:192.168.0.170:8002;nat=no;ftag=as14d7523e;lr=on
Max-Forwards: 70
From: "asterisk" sip:110@172.26.101.10:5080;tag=as14d7523e
To: sip:110@172.26.101.50:8002;tag=1749303708
Contact: sip:110@172.26.101.10:5080
Call-ID: 7e723f1c64086e964df79e493350a2a4@172.26.101.10:5080
CSeq: 102 ACK
User-Agent: ast01
Content-Length: 0
With code i posted before i have now issue to answer calls from ast (generated by asterisk)
200 From Phone arrives fine to Kamailio
<--- SIP read from UDP:172.26.101.50:8002 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.26.101.10:5080;rport=5080;branch=z9hG4bK125b3b98
Record-Route: sip:192.168.0.170:8002;nat=no;ftag=as14d7523e;lr=on
From: "asterisk" sip:110@172.26.101.10:5080;tag=as14d7523e
To: sip:110@172.26.101.50:8002;tag=1749303708
Call-ID: 7e723f1c64086e964df79e493350a2a4@172.26.101.10:5080
CSeq: 102 INVITE
Contact: sip:110@PUBLIC.IP:5066
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.37
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 255
v=0
o=110 8003 8000 IN IP4 172.26.101.41
s=SIP Call
c=IN IP4 172.26.101.41
t=0 0
m=audio 8424 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sdp_proxied:yes
<------------->
I see 200 OK sent to Asterisk
Asterisk sent to me ACK but Kamailio seems to do not send to Phone behind nat
3.- 192.168.0.170:8002
Thanks
2015-07-29 9:18 GMT+02:00 Alex Balashov abalashov@evaristesys.com:
Alberto,
- What are the literal (natively homed) IP addresses of Asterisk and
Kamailio?
- What is the Request Line (first line) of the ACK request being sent
from Asterisk, i.e.
ACK sip:... SIP/2.0
- To what IP and port is the ACK being sent by Asterisk?
-- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users