20 May
2006
20 May
'06
2:07 p.m.
What should I do? Something to set ? Why if I use the asterisk server
instead of the ser one and the same voip clients (using users with the nat
flag enabled) I can enstablish the communication without any problem ?
Should I open port 8000 on the router of the Lan ?
I 've attached the X-Lite Log when logging in :
---------------------
© 2005 CounterPath Solutions Inc. All rights reserved.
X-Lite release 1105x build stamp 21407
License key: B608ADE2A7C14E53835E7FF2EFAE95D0
Established SIP protocol listen on: 192.168.0.102:5060
NAT firewall type discovered is forced.
Discovered Symmetric NAT Firewall
SIP: 192.168.0.102:5060
RTP: 192.168.0.102:8000
NAT: 62.101.126.230
PROXY#0: 80.105.2.110:5060
SEND TIME: 1079327
SEND >> 80.105.2.110:5060
REGISTER sip:80.105.2.110 SIP/2.0
Via: SIP/2.0/UDP
62.101.126.230:5060;rport;branch=z9hG4bK01C14B200F494466A016FE0E0E46B658
From: claudio <sip:claudio@80.105.2.110>;tag=4251097143
To: claudio <sip:claudio@80.105.2.110>
Contact: "claudio" <sip:claudio@62.101.126.230:5060>
Call-ID: 4DAD46A40FE546019F15CFD05AAE23FD(a)80.105.2.110
CSeq: 9157 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1105x
Content-Length: 0
RECEIVE TIME: 1079425
RECEIVE << 80.105.2.110:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
62.101.126.230:5060;rport=39914;branch=z9hG4bK01C14B200F494466A016FE0E0E46B658
From: claudio <sip:claudio@80.105.2.110>;tag=4251097143
To: claudio
<sip:claudio@80.105.2.110>;tag=2a08c327b8d597781b526eaf86695180.4235
Call-ID: 4DAD46A40FE546019F15CFD05AAE23FD(a)80.105.2.110
CSeq: 9157 REGISTER
WWW-Authenticate: Digest realm="80.105.2.110",
nonce="446f5b8ae893e0b8c2a781a06a5257a485ebe0b6"
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 10.0.0.133:5060 "Noisy feedback tells: pid=13858
req_src_ip=62.101.126.230 req_src_port=39914 in_uri=sip:80.105.2.110
out_uri=sip:80.105.2.110 via_cnt==1"
SEND TIME: 1079427
SEND >> 80.105.2.110:5060
REGISTER sip:80.105.2.110 SIP/2.0
Via: SIP/2.0/UDP
62.101.126.230:5060;rport;branch=z9hG4bKF82CE6FE2C624791B7ABF06D7B8E7BFD
From: claudio <sip:claudio@80.105.2.110>;tag=4251097143
To: claudio <sip:claudio@80.105.2.110>
Contact: "claudio" <sip:claudio@62.101.126.230:5060>
Call-ID: 4DAD46A40FE546019F15CFD05AAE23FD(a)80.105.2.110
CSeq: 9158 REGISTER
Expires: 1800
Authorization: Digest
username="claudio",realm="80.105.2.110",nonce="446f5b8ae893e0b8c2a781a06a5257a485ebe0b6",response="91b05be86378bf515209116e96bd6314",uri="sip:80.105.2.110"
Max-Forwards: 70
User-Agent: X-Lite release 1105x
Content-Length: 0
RECEIVE TIME: 1079533
RECEIVE << 80.105.2.110:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP
62.101.126.230:5060;rport=39914;branch=z9hG4bKF82CE6FE2C624791B7ABF06D7B8E7BFD
From: claudio <sip:claudio@80.105.2.110>;tag=4251097143
To: claudio
<sip:claudio@80.105.2.110>;tag=2a08c327b8d597781b526eaf86695180.d540
Call-ID: 4DAD46A40FE546019F15CFD05AAE23FD(a)80.105.2.110
CSeq: 9158 REGISTER
Contact: <sip:claudio@62.101.126.230:5060>;expires=1800
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 10.0.0.133:5060 "Noisy feedback tells: pid=13858
req_src_ip=62.101.126.230 req_src_port=39914 in_uri=sip:80.105.2.110
out_uri=sip:80.105.2.110 via_cnt==1"
---------------------------------------
Hi,
probably the solution might be that your user agent in public will have to
re-register every minute or shorter time.
It is because the NAT binding between SER and the public user will be
closed if there is no SIP traffic.
Ladislav
Americania .it wrote:
I've installed ser 0.9.4 on Suse 10 and
it's running with mysql.
I can register users (I'm using serweb) but I can only make a call from a
user inside the Lan (where ser is installed) to a user outside behind a
Nat (Fastweb network in Italy, Natted). If i call from this user behind
Nat the call does not reach the other user. Viceversa it works fine.
I installed the nathelper module and rtpproxy too.
I don't know if I configured it well . Have I got to open some additional
port in my router on the lan where ser is running (I opened port 5060 udp
and tcp only)?
I've tried to use Asterix on another pc in the same lan and I don't have
this kind of problems with the some users and some clients (3cx phone).
Any suggestions ?
thanks
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