Dear All
i have a network , and all of nodes have valid ip addresses, i have
the ser 0.8.12 (i386/linux) with valid ip :
My clients -------------> SER box (sip proxy) ------------->
carrier (Cisco Box )
valid ip valid ip
valid ip
Windows messenger version: 0.8.12 IOS :12.2.4
The current config of my ser proxy is :
fifo="/tmp/ser_fifo"
alias=panaisp.net
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/auth_radius.so"
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser")
modparam("auth_radius", "radius_config",
"/usr/local/etc/radiusclient/radiusclient.conf")
modparam("auth_radius", "service_type", 15)
modparam("acc", "radius_config",
"/usr/local/etc/radiusclient/radiusclient.conf")
modparam("acc", "log_level", 1)
modparam("acc", "radius_flag", 1)
modparam("acc", "radius_missed_flag", 2)
modparam("acc", "log_fmt", "duprmiocfst")
modparam("acc", "early_media", 1)
route{
/* ********* ROUTINE CHECKS ********************************** */
if (!mf_process_maxfwd_header("10")) {
log("LOG: Too many hops\n");
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Wow -- Message too large");
break;
};
setflag(1);
setflag(2);
acc_rad_request("");
if (method=="INVITE") {
record_route();
if (!radius_www_authorize("panaisp.net")) {
www_challenge("panaisp.net", "0");
break;
};
}
if (uri==myself) {
if (method=="REGISTER") {
if (!radius_www_authorize("panaisp.net")) {
www_challenge("panaisp.net", "0");
break;
};
save("location");
break;
};
if (lookup("location")) {
forward(uri:host, uri:port);
break;
};
};
if (uri=~"^sip:[0-9]*@") {
if (uri=~"^sip:00" ) {
strip(2);
}
if (uri=~"^sip:[2-9]") {
prefix("82135011");
} else {
prefix("82135");
}
rewritehostport("63.243.143.8:5060");
};
forward(uri:host, uri:port);
}
I want to force the RTP traffic of my clients pass through the ser server .
Then i deside to use the RTP proxy + nathelper .
Please advise me that what change i must do in my current ser config ???
Best Regards
Ehsan
Hi,
Does anyone have any experience on building SER as an outbound proxy?
(i.e., the one that translates IP Addresses when the client is unable
to use STUN). An example config would be great.
Thank you,
Emilio Panighetti
Does SER support conferencing between user agents that
use different codecs (G.729, G.711), or does SIP deal
with those kinds of things differently?
Also, is there an auto-discovery/zeroconf protocol for
SIP user agents to automatically register newly
plugged in SIP devices?
How does SER compare to sipXproxy?
Luis.
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Hi,
Intel has an open source code for g729. This sample supports the ITU-T G.729
Recommendation and Annexes A, B, D and E.
Someone has adapted this with *. http://www.readytechnology.co.uk/open/g729/
Is it possible to integrate this with sems?
Also Intel has other sample code for g723, g728 and g722.
Thanks,
Richard
I have a problem when I run ser_mysql.sh
when I click enter after ¨Domain (realm) for the
default user 'admin':¨
it show me this message
¨creating database ser ...
/usr/local/sbin/ser_mysql.sh: line 80: mysql: command
not found¨
How can I solve this problem?
Thank....
__________________________________________________
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Can someone please explain why I am getting stale nonces? I am mostly
getting them with when a sipura spa-2000 tries to register. I get it
sometimes when using a granstream 100. Also, I am seeing 5 register
requests every time, is this normal?
Hello All.
I've been trying to find documentation on these parameters in my
<serweb>/config/config.php file.
Can someone either point me in the right direction for tell me what these
Click-To-Dial settings should be and what they are used for?
$config->ctd_target="sip:699@1.2.3.4";
$config->ctd_uri="sip:699@1.2.3.4";
Cheers,
Paul
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I have a routing problem which I cant explain.
A UA communicates with a Sip-Server over ser as a proxy. As the Sip-Server
only supports TCP connection the t_relay_to_tcp function is used. And as
the Sip-Server doesn't register to the Ser proxy a routeing for that
destination (sip server) is done by hand. See ser.cfg and Sip messages
below.
Everything seems to work just find except the BYE Message form the Sip
Server via the proxy to the UA. Ser sends the received BYE to himself
(Message 8) and then back to the Sip Server (Message 9).
listening points:
UA: 192.168.0.5:5555
ser: 192.168.0.5:5060
sipserver: 192.168.0.5:5080
Any suggestions?
ser.cfg
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (method=="INVITE") record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# dest SipAS? then relay via tcp without lookup
if(uri=~"sip:sipas@192.168.0.5" || uri=~"sip:192.168.0.5:5080") {
if (!t_relay_to_tcp("192.168.0.5","5080")) {
sl_reply_error();
break;
};
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
log("saving location \n");
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
log("not in locDB\n");
break;
};
};
if (!t_relay()) {
sl_reply_error();
break;
};
}
###########################################################################
Call flow:
1. ==== UA sends to ser
INVITE sip:sipas@192.168.0.5 SIP/2.0
Call-ID: f978201facd1ba9e7345af63683f96a1(a)192.168.0.5
CSeq: 1 INVITE
From: "user" <sip:user@192.168.0.5:5555;transport=udp>;tag=25591289
To: <sip:sipas@192.168.0.5>
Via: SIP/2.0/UDP
192.168.0.5:5555;branch=z9hG4bK98fa6f82a8447d74fe9a04da3d9dd0a1
Max-Forwards: 70
Contact: "user" <sip:192.168.0.5:5555;transport=udp>
Content-Type: application/sdp
Content-Length: 154
(sdp not shown)
---------------------------------------------
2. ==== ser to Sip Server
INVITE sip:sipas@192.168.0.5 SIP/2.0
Record-Route:
<sip:sipas@192.168.0.5;transport=tcp;r2=on;ftag=25591289;lr=on>
Record-Route: <sip:sipas@192.168.0.5;r2=on;ftag=25591289;lr=on>
Call-ID: f978201facd1ba9e7345af63683f96a1(a)192.168.0.5
CSeq: 1 INVITE
From: "user" <sip:user@192.168.0.5:5555;transport=udp>;tag=25591289
To: <sip:sipas@192.168.0.5>
Via: SIP/2.0/TCP 192.168.0.5;branch=z9hG4bK6446.87704592.0
Via: SIP/2.0/UDP
192.168.0.5:5555;branch=z9hG4bK98fa6f82a8447d74fe9a04da3d9dd0a1
Max-Forwards: 69
Contact: "user" <sip:192.168.0.5:5555;transport=udp>
Content-Type: application/sdp
Content-Length: 154
sdp not shown
---------------------------------------------
3. ==== Sip Server to ser
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.0.5;branch=z9hG4bK6446.87704592.0
Via: SIP/2.0/UDP
192.168.0.5:5555;branch=z9hG4bK98fa6f82a8447d74fe9a04da3d9dd0a1
Record-Route:
<sip:sipas@192.168.0.5;ftag=25591289;r2=on;lr=on;transport=tcp>
Record-Route: <sip:sipas@192.168.0.5;ftag=25591289;r2=on;lr=on>
From: user <sip:user@192.168.0.5:5555;transport=udp>;tag=25591289
To: <sip:sipas@192.168.0.5>;tag=swwmb9
Call-ID: f978201facd1ba9e7345af63683f96a1(a)192.168.0.5
CSeq: 1 INVITE
Contact: <sip:192.168.0.5:5080;transport=tcp>
Content-Length: 163
Content-Type: application/sdp
sdp not shown
---------------------------------------------
4. ==== ser to UA
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.5:5555;branch=z9hG4bK98fa6f82a8447d74fe9a04da3d9dd0a1
Record-Route:
<sip:sipas@192.168.0.5;ftag=25591289;r2=on;lr=on;transport=tcp>
Record-Route: <sip:sipas@192.168.0.5;ftag=25591289;r2=on;lr=on>
From: user <sip:user@192.168.0.5:5555;transport=udp>;tag=25591289
To: <sip:sipas@192.168.0.5>;tag=swwmb9
Call-ID: f978201facd1ba9e7345af63683f96a1(a)192.168.0.5
CSeq: 1 INVITE
Contact: <sip:192.168.0.5:5080;transport=tcp>
Content-Length: 163
Content-Type: application/sdp
sdpnot shown
---------------------------------------------
5. ==== UA to ser
ACK sip:192.168.0.5:5080;transport=udp SIP/2.0
Call-ID: f978201facd1ba9e7345af63683f96a1(a)192.168.0.5
CSeq: 1 ACK
From: "user" <sip:user@192.168.0.5:5555;transport=udp>;tag=25591289
To: <sip:sipas@192.168.0.5>;tag=swwmb9
Via: SIP/2.0/UDP 192.168.0.5:5555;branch=3139322e3136382e302e353a3535353
Max-Forwards: 70
Route: <sip:sipas@192.168.0.5;ftag=25591289;r2=on;lr=on>
Content-Length: 0
---------------------------------------------
6. ==== ser to Sip Server
ACK sip:192.168.0.5:5080;transport=udp SIP/2.0
Call-ID: f978201facd1ba9e7345af63683f96a1(a)192.168.0.5
CSeq: 1 ACK
From: "user" <sip:user@192.168.0.5:5555;transport=udp>;tag=25591289
To: <sip:sipas@192.168.0.5>;tag=swwmb9
Via: SIP/2.0/TCP 192.168.0.5;branch=0
Via: SIP/2.0/UDP 192.168.0.5:5555;branch=3139322e3136382e302e353a3535353
Max-Forwards: 69
Content-Length: 0
---------------------------------------------
7. ==== Sip Server to ser
BYE sip:user@192.168.0.5:5555;transport=udp SIP/2.0
Via: SIP/2.0/TCP 192.168.0.5:5080;branch=z9hG4bKf7kfy5
Route: <sip:sipas@192.168.0.5;ftag=25591289;r2=on;lr=on;transport=tcp>
Route: <sip:sipas@192.168.0.5;ftag=25591289;r2=on;lr=on>
Max-Forwards: 70
From: <sip:sipas@192.168.0.5>;tag=swwmb9
To: user <sip:user@192.168.0.5:5555;transport=udp>;tag=25591289
Call-ID: f978201facd1ba9e7345af63683f96a1(a)192.168.0.5
CSeq: 1 BYE
Content-Length: 0
---------------------------------------------
8. ==== ser to ser!
BYE sip:sipas@192.168.0.5;ftag=25591289;r2=on;lr=on SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bK6446.97704592.0;i=2
Via: SIP/2.0/TCP 192.168.0.5:5080;branch=z9hG4bKf7kfy5
Max-Forwards: 69
From: <sip:sipas@192.168.0.5>;tag=swwmb9
To: user <sip:user@192.168.0.5:5555;transport=udp>;tag=25591289
Call-ID: f978201facd1ba9e7345af63683f96a1(a)192.168.0.5
CSeq: 1 BYE
Content-Length: 0
---------------------------------------------
9. ==== ser to Sip Server! (and not to UA)
BYE sip:sipas@192.168.0.5;ftag=25591289;r2=on;lr=on SIP/2.0
Via: SIP/2.0/TCP 192.168.0.5;branch=z9hG4bK6446.a7704592.0
Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bK6446.97704592.0;i=2
Via: SIP/2.0/TCP 192.168.0.5:5080;branch=z9hG4bKf7kfy5
Max-Forwards: 68
From: <sip:sipas@192.168.0.5>;tag=swwmb9
To: user <sip:user@192.168.0.5:5555;transport=udp>;tag=25591289
Call-ID: f978201facd1ba9e7345af63683f96a1(a)192.168.0.5
CSeq: 1 BYE
Content-Length: 0
---------------------------------------------
10. ==== ser to Sip Server!
BYE sip:sipas@192.168.0.5;ftag=25591289;r2=on;lr=on SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bK6446.97704592.0;i=2
Via: SIP/2.0/TCP 192.168.0.5:5080;branch=z9hG4bKf7kfy5
Max-Forwards: 69
From: <sip:sipas@192.168.0.5>;tag=swwmb9
To: user <sip:user@192.168.0.5:5555;transport=udp>;tag=25591289
Call-ID: f978201facd1ba9e7345af63683f96a1(a)192.168.0.5
CSeq: 1 BYE
Content-Length: 0
and so on.....
Hi!
I have tried this with two diferents UA.... ATA -186 version 2.15 and
Sipura/SPA2000-2.0.10
and always the same result.... but one thing i want to point out:
the UA is not sending the ACK message only when it is located behind NAT and
Ser/rtp proxy are behind another NAT......if SER/Rtp Box have real ip, no
behind NAT then the user Agent is performing well and the call get two-way
audio....
How can't this change it's behavior....??? always the UA is behind NAT...
Always we're using rtpProxy....we modified the nathelper to include
Advertise address of the rtpProxy when we located SER/rtpProxy behind Nat...
and we can make successful calls from and to External customers with realip
both.... but if one is behind nat....then we get just one way audio, because
the UA behind nat is not sending ACK message...
Any clue???? Help!
Please... Thanks!
Armando
Cti, Miami, Fl
Hello Everyone.
I've got a glitch in my call waiting feature and was hoping someone could
suggest a way to fix it or debug it.
The Problem; I have three SIP phones. Two are Grandstream Budgetone 100's and
the third is a Cisco ATA 186. The firmware on the Grandstreams is version
1.0.5.11 and the ATA 186 is version 2.16.2
If a call is established between two of the phones and the third phone calls, I
do get the call waiting indicator. I can then press my "flash" button to take
the call. I can also press "flash" again to return to the original caller.
But if I press "flash" a third time to return to the new caller, audio can be
sent but not received. It's almost like one of the RTP channels has been
closed.
Could this be an error in my ser.cfg routing? I'm using ser-0.8.99-dev7
Regards,
Paul
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