+----------+
+-----+ +------+ |OpenSER | +------+
|Phone| |Router| |MediaProxy| |Router|
+-----+ +------+ +----------+ +------+
| | | | | |
--------------------- -------------------------- -------------- Internet
Private LAN A Private LAN B
Here is the situation : SIP phones are in a private LAN A. They can speak
with the server
hosting openser and mediaproxy which is located on private LAN B. This
server can speak
with the internet through a router that nat (1:1) its private IP on a public
one.
To be able to interconnect inside sip phone with outside ones, openser must
advertise the
private address of the mediaproxy to the inside sip phones and its public
(nated) address
to the outside sip phones. But afaik , the media proxy module doesn't let me
dynamically
choose the IP which replace the phone IP in the SDP.
How do you think I can get through this ?
--
Simon Morvan.
Hi,
Does someone have an axample of using avp_db_query() with a mysql stored
procedure? I keep getting an error "submit_query: PROCEDURE
openser.sp_getrouteinfo can't return a result set in the given context"
whenever I pass a stored procedure call to that functions. Thanks.
Mirek N
Globalive Communications Corp.
The Future of Voice
T:416-640-1088
F:416-640-1089
60 Adelaide St E, 6th Floor
Toronto, ON, M5C 3E4
www.globalive.com
All,
OpenSER Admin version 0.3 has been released. This fixes the bug where
compatibility with openserctl was broken, and adds the ability to add other
management accounts to the interface.
Let me know what you think.
Thanks,
Mike Williams
Hi!
Description of the problem:
The some UA can send only IP addresses as domains.
Some subscribers incorrectly configure their UA and as a result as the
domain IP of SIP Proxy comes.
In result in the table location the part of subscribers has the
symbolical domain (voapp.ru) and part - IP address of SIP Proxy
(62.33.22.14)
It leads to to the certain complexities in switching incming calls.
The problem disappears if to tell:
modparam("usrloc", "use_domain", 0)
modparam("registrar", "use_domain", 0)
And use only names without domains.
But in our case it does not approach since we have some independent
domains.
I tried to rewrite fields From and To
route(8);
save("location");
route[8] {
#correct From and To in REGISTER messages with IP address instead of
domain name
if (search("From:.*@[0-9]+\.[0-9]+\.[0-9]+\.[0-9]+.*") or
search("To:.*@[0-9+\.[0-9]+\.[0-9]+\.[0-9]+.*")) {
subst("/^From:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/Fr
om:\1sip:\2@voapp.ru\7/i");
subst("/^To:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/To:\
1sip:\2@voapp.ru\7/i");
xlog("L_ERR", "REGISTER From To done\n");
return(-1);
};
return(1);
}
Fields varied but function save took their not changed values from
initial message.
After that I have tried to send REGISTER message once again to myself
for processing:
if (!route(8)) {
t_relay();
} else {
save("location");
};
In this case in the location table there was a record with the corrected
name of the domain.
But in Contact field there was an address of SIP Proxy server instead of
address of UA.
Can anybody prompt idea how to to change domain in From To fields at
reception of REGISTER message?
Tnx.
How to change domain when REGISTERing UA?
Hi!
Description of the problem:
The some UA can send only IP addresses as domains.
Some subscribers incorrectly configure their UA and as a result as the
domain IP of SIP Proxy comes.
In result in the table location the part of subscribers has the
symbolical domain (voapp.ru) and part - IP address of SIP Proxy
(62.33.22.14)
It leads to to the certain complexities in switching incming calls.
The problem disappears if to tell:
modparam("usrloc", "use_domain", 0)
modparam("registrar", "use_domain", 0)
And use only names without domains.
But in our case it does not approach since we have some independent
domains.
I tried to rewrite fields From and To
route(8);
save("location");
route[8] {
#correct From and To in REGISTER messages with IP address instead of
domain name
if (search("From:.*@[0-9]+\.[0-9]+\.[0-9]+\.[0-9]+.*") or
search("To:.*@[0-9+\.[0-9]+\.[0-9]+\.[0-9]+.*")) {
subst("/^From:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/Fr
om:\1sip:\2@voapp.ru\7/i");
subst("/^To:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/To:\
1sip:\2@voapp.ru\7/i");
xlog("L_ERR", "REGISTER From To done\n");
return(-1);
};
return(1);
}
Fields varied but function save took their not changed values from
initial message.
After that I have tried to send REGISTER message once again to myself
for processing:
if (!route(8)) {
t_relay();
} else {
save("location");
};
In this case in the location table there was a record with the corrected
name of the domain.
But in Contact field there was an address of SIP Proxy server instead of
address of UA.
Can anybody prompt idea how to to change domain in From To fields at
reception of REGISTER message?
Tnx.
Dmitry
Thanks...found it 2 seconds after sending the email! :PDate: Thu, 24 Aug 2006 21:21:10 +0200From: greger(a)teigre.comTo: steve_helfen(a)hotmail.comCC: serusers(a)iptel.orgSubject: Re: [Serusers] Alternate port for rewritehost
Use rewritehostport("domain:5060")
g-)
Stephen Helfen wrote:
All,
I am receiving SIP traffic on a non-5060 port but need to change the
port of the rewritehost to 5060 as the carrier is dropping my INVITE
messages with the non-5060 port. Any suggestions?
Thanks,
Steve
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Hi all:
The document metion that "RTPproxy will not be installed on the SER
server, but on a remote machine." If I install SER on "192.168.1.3", and
install RTPproxy on "192.168.1.5". How do they work?
Now, I install they on a same machine, and I run SER first, then usr
command "rtpproxy -l xxx.xxx.xxx.xxx" to force rtpproxy. The SIP client can
register behind NAT, but there are some problem with RTP stream. I found
that RTP strem is able to relay to this machine, but this machine do not
transmit those data to destination!!
PC(A)
Server(SER+RTPproxy) PC(B)
RTP
Stream(A) RTP Stream(B)
1. ---------------------------->
<----------------------------
2. Receive RTP stream A and B
3. Server should transmit RTP stream(A) to PC(B) and RTP stream(B) to
PC(A), but this step do not work. (I am not sure that server does not
forward the RTP packets or Clients do not receive RTP packet.)
What is the problem? How should I operate if RTPproxy install on another
machine? Because the command "rtpproxy" will not be used on the machine
which have installed SER.
Please help me, thank you.
Regards,
Caxton
All,I am receiving SIP traffic on a non-5060 port but need to change the port of the rewritehost to 5060 as the carrier is dropping my INVITE messages with the non-5060 port. Any suggestions?Thanks,Steve
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Dear all,
Thank you very much to all of you for help me to answer my question about serweb before.
So, based on your suggestion, I tried to install serweb 0.9.3 for my openser server.
But, when I tried to install it, I am confused. I don`t understand about the installation note that I have got from untar the serweb 0.9.3 packet.
These are some questions that I need to ask:
1. Where should we untar the serweb 0.9.3 packet? Should we untar it in root or anywhere we like?
Because when we untar it, there will be a serweb folder in the location where we untar it (for example, when we untar the packet in root there will be a serweb directory in root) . This serweb folder also contain html directory and phplib directory.
As I know, There are also html directory in /root/var/www.
So, What directory should I use? Should I delete the html directory and phplib directory in /root/var/www/?
2. Is there any relationship between the html directory in /root/var/www and html directory from /root/serweb?
Which one should I use?
3. In the installation note I got this message :
"install it, unpack the distibution on your webserver so that the directory html is in document root and phplib is not accessible via web"
Can anybody help me to explain what this message means? and give me an example?
4. If I can make a wish for all of you, I do hope anybody can share the serweb configuration that she/he has configured to me. So, it can become the guidelines for me.
That `s all the question that I got when I tried to install the serweb. I am confused. I do hope anybody can help me because I am new in this system.
Thank you very much for your help. Thanks
Regards,
Ferianto
---------------------------------
Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.
I'm testing my rtpproxy config with a linksys phone and a grandstream phone,
either not NAT'ed, or behind a cisco (full cone), or behind a Netgear (symmetric).
Things generally work, but I have a problem with the grandstream
behind the symmetric nat.
This is the register:
U NAT-IP:3284 -> PROXY-IP:5060
REGISTER sip:sip.example.com SIP/2.0
Via: SIP/2.0/UDP NAT-IP:3280;branch=z9hG4bK0103d0ff79cc9c02
From: "Grandstream" <sip:frick@sip.example.com;user=phone>;tag=743335557afcade4
To: <sip:frick@sip.example.com;user=phone>
Contact: <sip:frick@NAT-IP:3280;user=phone>
Authorization: Digest username="frick", realm="sip.example.com", algorithm=MD5, uri="sip:sip.example.com", nonce="foo", response="bar"
Call-ID: 11f6defb0857db17(a)192.168.0.100
CSeq: 116 REGISTER
Expires: 60
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
This is the resulting contact:
...Record(0x2864b680)..
domain: 'location'
aor : 'frick(a)sip.example.com'
~~~Contact(0x2864d4e0)~~~
domain : 'location'
aor : 'frick(a)sip.example.com'
Contact : 'sip:frick@NAT-IP:3280;user=phone'
Expires : 34
q :
Call-ID : '11f6defb0857db17(a)192.168.0.100'
CSeq : 119
User-Agent: 'Grandstream BT100 1.0.6.7'
received : 'sip:NAT-IP:3284'
Path : ''
State : CS_SYNC
Flags : 1
Sock : PROXY-IP:5060 (0x8122f50)
Methods : 5695
next : 0x0
prev : 0x0
~~~/Contact~~~~
.../Record..
Now, what I don't understand is the two ports used above: 3280 and
3284. It looks like the phone thinks it is being natted to port 3280,
but it's really getting port 3284
When I make a call I get
U NAT-IP:3284 -> PROXY-IP:5060
INVITE sip:16509415674@sip.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP NAT-IP:3280;branch=z9hG4bK4a4c039fbcb5bef8
From: "Grandstream" <sip:frick@sip.example.com;user=phone>;tag=2dc2fc3c94e93951
To: <sip:16509415674@sip.example.com;user=phone>
Contact: <sip:frick@NAT-IP:3280;user=phone>
Supported: replaces
Call-ID: 1f04f9c6ec474f70(a)192.168.0.100
CSeq: 26381 INVITE
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 419
v=0
o=frick 8000 8002 IN IP4 NAT-IP
s=SIP Call
c=IN IP4 NAT-IP
t=0 0
m=audio 3295 RTP/AVP 0 8 4 18 2 15 97 9 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/16000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
and my side sends back the Proxy Authentication Required, but it sends
it to port 3280. Now, is that based on the Via: header or the Contact:
header, or (unlikely) the stored contact info from the REGISTER?
The phone acts like it never got it (and I'm guessing it didn't) and
sends the INVITE again without any auth data.
The thing that bugs me is that this is an outbound call, from the
phone out through the NAT box, I figured that would be doable.
The Linksys942 works, so I'm hoping this is a Grandstream bug,
or something I can fix in my openser config.
Thanks,
-mark