Hi Javier,
Thanks for your information. I need to make SEP to
send/proxy the message to my presence server. SEP and presence server
are physically different entities. Would be of greater help if you could
let me know how can i specify the IP address of my presence server in
SEP so that the presence specific messages reaching SEP gets forwarded
to my presence server.
Thanks and Regards,
Senthil Kumar,
________________________________
From: Javier Ramirez [mailto:jvr_78@yahoo.com.ar]
Sent: 25 September 2006 15:03
To: Nachimuthu,SK,Senthil,XSD4C C
Subject: Re: [Serusers] Presence
if (method=="SUBSCRIBE") {
if (!t_newtran()) {
sl_reply_error();
};
handle_subscription("registrar");
};
if (method=="PUBLISH") {
if (!t_newtran()) {
log(1, "newtran error\n");
sl_reply_error();
};
handle_publish("registrar");
break;
};
----- Original Message -----
From: senthil.nachimuthu(a)bt.com
To: serusers(a)lists.iptel.org
Sent: Monday, September 25, 2006 8:28 AM
Subject: [Serusers] Presence
Hi,
I want to configure SER <file://ser/about> 0.9.6 in such
a way that It can proxy presence messages(SUBSCRIBE,PUBLISH) to presence
server. Can you let me know how can I do that?
Thanks and Regards,
Senthil Kumar,
Presence Capability,
BT-Service Execution Platform,
B62-MH,Orion Building,Adastral Park,Ipswich
Work:+44 (0)1473 605 668
Mobile:+44 (0)7772287447
__________ NOD32 1.1768 (20060922) Information __________
This message was checked by NOD32 antivirus system.
http://www.eset.com
________________________________
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
__________ NOD32 1.1768 (20060922) Information __________
This message was checked by NOD32 antivirus system.
http://www.eset.com
If I use
# -- nathelper --
modparam("nathelper", "rtpproxy_sock", "udp:59.124.210.41:22222") #
default port 22222
#modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_from", "sip:59.124.210.41")
# -----------------------------------------------------------------
# NAT detection
# -----------------------------------------------------------------
route(2);
route[1] {
if (subst_uri('/(sip:.*);nat=yes/\1/')){
setflag(6);
};
if (isflagset(5)||isflagset(6)) {
route(3);
}
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[2]{
force_rport();
if (nat_uac_test("19")) {
if (method=="REGISTER") {
fix_nated_register();
} else {
fix_nated_contact();
};
setflag(5);
};
}
route[3] {
if (is_method("BYE|CANCEL")) {
unforce_rtp_proxy();
} else if (is_method("INVITE")){
force_rtp_proxy();
t_on_failure("1");
};
if (isflagset(5))
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
t_on_reply("1");
}
failure_route[1] {
if (isflagset(6) || isflagset(5)) {
unforce_rtp_proxy();
}
}
I get in the log file:
Oct 4 09:19:01 taipei Taipei-openser[4083]: WARNING: rtpp_test: support
for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily
Oct 4 09:19:01 taipei Taipei-openser[4085]: WARNING: rtpp_test: support
for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily
Oct 4 09:19:01 taipei Taipei-openser[4087]: WARNING: rtpp_test: support
for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily
Oct 4 09:19:01 taipei Taipei-openser[4089]: WARNING: rtpp_test: support
for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily
Oct 4 09:19:01 taipei Taipei-openser[4091]: WARNING: rtpp_test: support
for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily
Oct 4 09:19:01 taipei Taipei-openser[4050]: WARNING: rtpp_test: support
for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily
Oct 4 09:19:01 taipei Taipei-openser[4067]: WARNING: rtpp_test: support
for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily
Oct 4 09:19:01 taipei Taipei-openser[4069]: WARNING: rtpp_test: support
for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily
with:
modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")
this error does not appear. In neither way it work for me (ring but no
audio)
Can anybody help me please?
bye
Ronald Wiplinger
We have STUN, nathelper with rtpproxy or mediaproxy, ... maybe even more.
Which one is good for which situation?
What are the differences?
How can I cover each situation?
Can I combine different methods?
bye
Ronald
Hi,
I have one SER and two asterisk boxes. I'm trying to duplicate each register
message from the client so it will
reach each of the asterisks. I tried the following in my openser.cfg:
if ( (is_method("SUBSCRIBE")) || (is_method("REGISTER")) ) {
seturi("sip:nobody@10.0.0.1:5060");
append_branch("sip:parallel@10.0.0.2:5060");
if (!t_relay()) {
sl_reply_error();
};
exit;
}
The first asterisk server registers the UA successfully. The second one
gives this error:
Oct 3 12:55:48 NOTICE[32364]: chan_sip.c:6395 check_auth: stale nonce
received from 'sip:1234566@foo.bar'
Transmitting (no NAT) to asterisk2:5060:
I googled a bit and I understand I cannot fully duplicate a sip message
since the nonce cannot be identical. Is there a way to recreate the sip
message with another nonce? Is there any other solution to this scenario?
Thanks,
Adi.
Does SER support the SRV mechanism, sending a request to a URL with a
SRV domain name?
Also, if the first priority destination fails due to a timeout, will it
failover to the second priority?
Thanks for help,
Paul
Hi,
Thanx for the answer last time...
I have add configuration of MSILO module in openser.cfg.
The problem is I don't know how to test it (how to leave offline voice message between clients). If there is any application I have to use to save voice message, please let me know.
And about the silo table, I've checked OPENSER database and I 've found that silo table is already there without I've to create it.
But, when I configure my serweb, I found in my /var/www/html/serweb/config.php,
there is another table named "voice_silo". There is no table named "voice_silo" in my OPENSER database. I'd like to know what's the different between "silo" and "voice_silo" (maybe in storing message (voice or text)).
Thanx
Regards,
Aldi
---------------------------------
Do you Yahoo!?
Everyone is raving about the all-new Yahoo! Mail.
Is it possible to set up SER with asterisk so that any INVITE that is
forwarded from SER to asterisk does not need to be authenticated by
asterisk?
Thanks,
Mark
Hi,
I want to get a success feeling, ... and with your help I would get my
openser installation running with:
a. mysql (done)
b. nat traversal (not working yet)
c. ENUM (not working yet)
d. sipbroker (not started yet) e.g., *777xxxxx should go to Sipgate.de ....
If you want to help me to set it up, please send me first an email and I
am happy to reserve time for your set-up try. If you succeed I pay you
US$ 20 to your PayPal account !!!!
Environment:
1. Eyebeam is on my internal network
2. several video phones are available outside of my network (one on a
public IP: Grandstream 3000 video phone) and others behind a nat
3. several ENUM numbers are available, which are going to my asterisk
server (different machine, different domain name)
4. sipbroker can also reach our asterisk server
71xx IP range 192.168.10.x
--------------
NAT 7111 (public IP)
|
|
---------------------------------------INTERNET
--------------------------------------------
|
|
|
extern: 203.70.36.126
|
|
second ADSL on
192.168.250.250
| public ip
address
|
taipei.elmit.net (Openser server)
other servers with 2 Ethernet |
| IP range
192.168.250.x | |
| | |
---------------------------------------------------------------------------------------------
| | | |
phones and desk top computers
bye
Ronald Wiplinger
I try to call an ENUM number. I use this settings for it:
mpath="/usr/local/lib64/openser/modules/"
loadmodule "enum.so"
# -- enum params --
modparam("enum", "domain_suffix", "e164.arpa.")
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");
exit;
};
if (msg:len >= 4096 ) {
sl_send_reply("513", "Message too big");
exit;
};
# -----------------------------------------------------------------
# NAT detection
# -----------------------------------------------------------------
log("I am checking for NAT now\n");
route(2);
# -----------------------------------------------------------------
# if you want to make ENUM work with numbers starting with "00",
# use the following to convert "00" it into a "+"
# -----------------------------------------------------------------
log("I am checking for ENUM now\n");
if (uri=~"^sip:00[1-9][0-9]*@") { # tried before:
taipei\.elmit\.net") {
# strip leading "00"
# (change example.net to your domainname or skip the stuff after
the "@")
strip(2);
# (adjust, if your international prefix is something else than "00")
prefix("+");
};
# check if request uri starts with an international phone
# number (+X.), if yes, try to ENUM resolve in e164.arpa.
# if no result, try in nrenum.net
if (uri=~"sip:\+[0-9]+@") { # tried before:
taipei\.elmit\.net") {
# (change example.net to your domainname or skip the stuff after
the "@")
####if ( !enum_query("e164.arpa.") ) {
#### enum_query("nrenum.net.");
#### };
if (enum_query()) {
t_relay();
exit;
}
};
I tried to add some log statements, but I never see them in
/var/log/messages either.
Below is the ngrep output. It seems that it does not go anywhere in route()
Any ideas how to track it down?
bye
Ronald
Content-Type: application/sdp.
User-Agent: eyeBeam release 1005g stamp 32576.
Content-Length: 809.
.
v=0.
o=- 2 2 IN IP4 192.168.250.108.
s=CounterPath eyeBeam 1.5.
c=IN IP4 192.168.250.108.
t=0 0.
m=audio 48702 RTP/AVP 107 100 106 6 0 105 18 3 5 101.
a=alt:1 2 : O5/HQNzE iLXVkGqH 5.32.86.166 48702.
a=alt:2 1 : ok7vH1EF jO6dCAlm 192.168.250.108 48702.
a=fmtp:18 annexb=yes.
a=fmtp:101 0-15.
a=rtpmap:107 BV32/16000.
a=rtpmap:100 SPEEX/16000.
a=rtpmap:106 SPEEX-FEC/16000.
a=rtpmap:105 SPEEX-FEC/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=sendrecv.
a=x-rtp-session-id:E4797B0FC1A140648811BD63523C3B70.
m=video 40488 RTP/AVP 125.
a=alt:1 2 : urVBRqXw O9lI2kQp 5.32.86.166 40488.
a=alt:2 1 : LCXVKz4X nxDzxq4V 192.168.250.108 40488.
a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800.
a=rtpmap:125 H264/90000.
a=sendrecv.
a=x-rtp-session-id:B8E2EE0EA3944D6DA47E5D98AABCD52B.
#
U 2006/09/28 16:57:11.567171 192.168.250.41:5060 -> 192.168.250.108:40430
SIP/2.0 404 Not Found.
Via: SIP/2.0/UDP
192.168.250.108:40430;branch=z9hG4bK-d87543-0c396723de34cf5a-1--d87543-;rport=40430.
To:
"00886228357765"<sip:00886228357765@192.168.250.41>;tag=329cfeaa6ded039da25ff8cbb8668bd2.e6c9.
From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=3f7a5072.
Call-ID: YmQ0YzNiYmU4MDJjMjQ4YzAyZjJjMjZiYzA3ZGYxYTk..
CSeq: 1 INVITE.
Server: OpenSer (1.1.0-notls (x86_64/linux)).
Content-Length: 0.
Warning: 392 192.168.250.41:5060 "Noisy feedback tells: pid=11197
req_src_ip=192.168.250.108 req_src_port=40430
in_uri=sip:00886228357765@192.168.250.41
out_uri=sip:+886228357765@192.168.250.41 via_cnt==1".
.
#
U 2006/09/28 16:57:11.569258 192.168.250.108:40430 -> 192.168.250.41:5060
ACK sip:00886228357765@192.168.250.41 SIP/2.0.
Via: SIP/2.0/UDP
192.168.250.108:40430;branch=z9hG4bK-d87543-0c396723de34cf5a-1--d87543-;rport.
To:
"00886228357765"<sip:00886228357765@192.168.250.41>;tag=329cfeaa6ded039da25ff8cbb8668bd2.e6c9.
From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=3f7a5072.
Call-ID: YmQ0YzNiYmU4MDJjMjQ4YzAyZjJjMjZiYzA3ZGYxYTk..
CSeq: 1 ACK.
Content-Length: 0.
.
Hi,
Can I change the authentication protocol for any other or the Openser authenticate only with the digest?
The Digest needs that the password in the DB stay in clear text and its not good.
Thanks.
---------------------------------
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