[OpenSIPS-Users] NAT type
Hola, como puedo detectar cual el tipo de NAT de un cliente? Saludos ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem NAT RTPproxy
In ngrep traffic check no active rdp-session-id
but do not know how to solve
#
U +3.135110 IP-ASTERISK:5060 -> IP_OPENSIPS:5060
INVITE sip:100@ IP_OPENSIPS SIP/2.0
Via: SIP/2.0/UDP IP-ASTERISK:5060;branch=z9hG4bK3e684698;rport
Max-Forwards: 70
From: "3414741468"
;tag=as33306c2a
To:
Contact:
Call-ID: 46ea6e9819e3583c59479d9304cc2b4f@IP-ASTERISK
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.20
Date: Mon, 26 Mar 2012 16:29:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 1324806659 1324806659 IN IP4 IP-ASTERISK
s=Asterisk PBX 1.6.2.20
c=IN IP4 IP-ASTERISK
t=0 0
m=audio 10788 RTP/AVP 0 18 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
tanks
Bogdan-Andrei Iancu wrote:
Well, you know, one is what we want to do , another we actually
get.
I was rather asking if, making a sip capture (with ngrep) you see
in your call the RTPproxy insertion - check it in traffic, not in
script.
Regards,
Bogdan
On 04/02/2012 10:05 PM, magnusadil...@gmail.com
wrote:
hi, yes, rtpproxy is active in invite 200
onreply_route[3] {
if ((isflagset(5) || isbflagset(0)) && status =~
"(183)|(2[0-9][0-9])" && has_body("application/sdp")) {
if (rtpproxy_answer()) {
log("L_INFO: rtpproxy_answer NAT");
}
}
if (!subst_uri('/(sip:.*);nat=yes/\1/'))
{
search_append('Contact:.*sip:[^>[:cntrl:]]*',
';nat=yes');
}
exit;
}
But i'm implemented this in invite route
if (is_method("INVITE") {
if ($si == "IP ASTERISK" && is_method("INVITE")) {
fix_nated_contact();
fix_nated_sdp("1");
xlog("L_INFO", "NAT detected3 PSTN for SIP");
setflag(5);
return;
}
}
and worked, but
I think it is not correct
tansk
Bogdan-Andrei Iancu wrote:
Hi Magnus,
attaching cfg files is useless, as no one will debug the
script, but we will help you to debug your script.
So, for the non-working case (PSTN to SIP) does your script
force RTPproxy in INVITE and 200 OK ?
Regards,
Bogdan
On 03/29/2012 01:52 AM, magnusadil...@gmail.com
wrote:
I have phones (some behind NAT) connecting to Opensips
server an Asterisk and an rtpproxy as seen below:
rtpproxy started with
ps -aux | grep rtpproxy
root 15666 0.0 0.0 14472 920 ? Ssl Mar23
0:05 ./rtpproxy -F -l 189.254.2.19 -s udp:* 7890 -d DBUG
LOG_LOCAL3
UAC1 username =
100Firewall/routerOpensips
1.7-- RTP PROXYAsterisk 1.6
192.168.1.10
192.168.1.1 65.254.63.212
189.254.2.19 190.61.201.89
external ip dinamic 169.254.2.2
- Calls between UAC are OK (both SIP and RTP).
- Calls UAC for PSTN is OK.
- Did numbers is received in Asterisk, and destination for
UAC registered in opensips, but no work audio .
(EX User call cellphone for DID 54115368566, call is
received in asterisk, and destination for user 100,
registered in opensips)
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--
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OpenSIPS Founder and Developer
http://www.opensips-solutions
Re: [OpenSIPS-Users] Problem NAT RTPproxy
hi, yes, rtpproxy is active in invite 200
onreply_route[3] {
if ((isflagset(5) || isbflagset(0)) && status =~
"(183)|(2[0-9][0-9])" && has_body("application/sdp")) {
if (rtpproxy_answer()) {
log("L_INFO: rtpproxy_answer NAT");
}
}
if (!subst_uri('/(sip:.*);nat=yes/\1/'))
{
search_append('Contact:.*sip:[^>[:cntrl:]]*',
';nat=yes');
}
exit;
}
But i'm implemented this in invite route
if (is_method("INVITE") {
if ($si == "IP ASTERISK" && is_method("INVITE")) {
fix_nated_contact();
fix_nated_sdp("1");
xlog("L_INFO", "NAT detected3 PSTN for SIP");
setflag(5);
return;
}
}
and
worked, but I think it
is not correct
tansk
Bogdan-Andrei Iancu wrote:
Hi Magnus,
attaching cfg files is useless, as no one will debug the script,
but we will help you to debug your script.
So, for the non-working case (PSTN to SIP) does your script force
RTPproxy in INVITE and 200 OK ?
Regards,
Bogdan
On 03/29/2012 01:52 AM, magnusadil...@gmail.com
wrote:
I have phones (some behind NAT) connecting to Opensips server an
Asterisk and an rtpproxy as seen below:
rtpproxy started with
ps -aux | grep rtpproxy
root 15666 0.0 0.0 14472 920 ? Ssl Mar23 0:05
./rtpproxy -F -l 189.254.2.19 -s udp:* 7890 -d DBUG LOG_LOCAL3
UAC1 username =
100Firewall/routerOpensips
1.7-- RTP PROXYAsterisk 1.6
192.168.1.10 192.168.1.1
65.254.63.212 189.254.2.19 190.61.201.89
external ip dinamic 169.254.2.2
- Calls between UAC are OK (both SIP and RTP).
- Calls UAC for PSTN is OK.
- Did numbers is received in Asterisk, and destination for UAC
registered in opensips, but no work audio .
(EX User call cellphone for DID 54115368566, call is received in
asterisk, and destination for user 100, registered in opensips)
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Problem NAT RTPproxy
I have phones (some behind NAT) connecting to Opensips server an
Asterisk and an rtpproxy as seen below:
rtpproxy started with
ps -aux | grep rtpproxy
root 15666 0.0 0.0 14472 920 ? Ssl Mar23 0:05
./rtpproxy -F -l 189.254.2.19 -s udp:* 7890 -d DBUG LOG_LOCAL3
UAC1 username =
100Firewall/routerOpensips
1.7-- RTP PROXYAsterisk 1.6
192.168.1.10 192.168.1.1
65.254.63.212 189.254.2.19 190.61.201.89
external ip dinamic 169.254.2.2
- Calls between UAC are OK (both SIP and RTP).
- Calls UAC for PSTN is OK.
- Did numbers is received in Asterisk, and destination for UAC
registered in opensips, but no work audio .
(EX User call cellphone for DID 54115368566, call is received in
asterisk, and destination for user 100, registered in opensips)
loadmodule "db_mysql.so"
loadmodule "signaling.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "mi_fifo.so"
loadmodule "uri.so"
loadmodule "acc.so"
loadmodule "dialog.so"
loadmodule "load_balancer.so"
loadmodule "nathelper.so"
loadmodule "siptrace.so"
loadmodule "rtpproxy.so"
loadmodule "auth.so"
loadmodule "auth_db.so"
loadmodule "domain.so"
modparam("load_balancer", "db_url",
"mysql://opensips:opensips@localhost/opensips")
modparam("siptrace", "db_url",
"mysql://opensips:opensips@localhost/opensips")
modparam("siptrace", "trace_flag", 22)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "enable_ack_trace", 1)
modparam("rtpproxy", "rtpproxy_sock","udp:189.254.2.19:7890")
modparam("dialog", "db_mode", 1)
modparam("dialog", "db_url",
"mysql://opensips:opensips@localhost/opensips")
modparam("nathelper", "natping_interval", 10)
modparam("nathelper", "natping_processes", 3)
modparam("nathelper", "natping_socket", "189.254.2.19:5006")
modparam("nathelper", "received_avp", "$avp(42)")
modparam("nathelper", "force_socket", "189.254.2.19:3")
modparam("nathelper", "sipping_from", "sip:pinger@65.254.63.212")
modparam("nathelper", "sipping_method", "INFO")
modparam("nathelper", "sipping_bflag", 7)
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("rr","enable_double_rr",1)
modparam("rr","append_fromtag",1)
modparam("registrar", "max_contacts", 10)
modparam("usrloc", "db_mode", 2)
modparam("usrloc",
"db_url","mysql://opensips:opensips@localhost/opensips")
modparam("acc", "early_media", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "detect_direction", 0)
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "password_column_2", "ha1b")
modparam("auth_db", "calculate_ha1", 1)
modparam("auth_db",
"db_url","mysql://opensips:opensips@localhost/opensips")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "skip_version_check", 1)
modparam("domain",
"db_url","mysql://opensips:opensips@localhost/opensips")
modparam("domain", "db_mode", 1) # Use caching
modparam("auth_db|usrloc|uri", "use_domain", 0)
route {
if (!mf_process_maxfwd_header("256")) {
if (method != "ACK") {
sl_send_reply("483", "Too Many Hops");
}
return;
}
if (msg:len > max_len) {
if (method != "ACK") {
sl_send_reply("513", "Message Overflow");
}
return;
}
if (status == "482") { #loop detection
xlog("L_INFO", "Webur: $mi $rm $fu -> $ru status 482 Loop
Detected\n");
return;
}
if (!mf_process_maxfwd_header("3")) {
sl_send_reply("483", "looping");
exit;
}
if (has_totag()) {
loose_route();
t_relay();
exit;
}
if (method == "INVITE") {
route(3);
return;
} else
if (method == "ACK") {
route(9);
return;
} else
if (method == "BYE" || method == "CANCEL") {
route(5);
return;
} else
Re: [OpenSIPS-Users] Users Digest, Vol 41, Issue 52
Buenas tardes, alguien de la lista habla español o portugues, y que este disponible para hacerme un pequeño trabajo en opensips? disculpe pero no hablo ingles saludos users-requ...@lists.opensips.org escreveu: Send Users mailing list submissions to users@lists.opensips.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.opensips.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-requ...@lists.opensips.org You can reach the person managing the list at users-ow...@lists.opensips.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Re: how does OpenSIPS manage 183's message?? (spady) 2. Re: Siptraces not shown on OpenSIPS-CP (spady) 3. Re: uac_replace_from corruption (Jeff Pyle) 4. SIP/SIMPLE to XMPP Gateway for SMS (DMF) 5. Re: B2BUA Ripping/Truncating Callid (Logan) -- Message: 1 Date: Thu, 15 Dec 2011 06:21:01 -0800 (PST) From: spady Subject: Re: [OpenSIPS-Users] how does OpenSIPS manage 183's message?? To: users@lists.opensips.org Message-ID: <1323958861503-7097238.p...@n2.nabble.com> Content-Type: text/plain; charset=us-ascii Hi Denis, that was it!!! It was setted to "auto" . I set it to "none" and now it works as aspected Perfet. Thank you very much for your hint ;-) Best regards -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/how-does-OpenSIPS-manage-183-s-message-tp7090462p7097238.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. -- Message: 2 Date: Thu, 15 Dec 2011 08:14:30 -0800 (PST) From: spady Subject: Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP To: users@lists.opensips.org Message-ID: <1323965670161-7097614.p...@n2.nabble.com> Content-Type: text/plain; charset=us-ascii Can someone help me with this? I checked again config and seems ok but form CP nothing yet. Regards -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Siptraces-not-shown-on-OpenSIPS-CP-tp7052741p7097614.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. -- Message: 3 Date: Thu, 15 Dec 2011 12:27:38 -0500 From: Jeff Pyle Subject: Re: [OpenSIPS-Users] uac_replace_from corruption To: OpenSIPS users mailling list Message-ID: Content-Type: text/plain; charset="iso-8859-1" Brett, Is the other end an Acme? If so, they need to implement some custom parameters (which I do not have) to* *honor some parts of section 12 of RFC3261 in such a way that won't break uac_replace_from(). Let me know if this is the case and we'll talk more. Rasvan, Can you share more about the "new" way to do it with the dialog module? Is this available in 1.6? - Jeff -- next part -- An HTML attachment was scrubbed... URL: -- Message: 4 Date: Thu, 15 Dec 2011 10:03:47 -0800 (PST) From: DMF Subject: [OpenSIPS-Users] SIP/SIMPLE to XMPP Gateway for SMS To: users@lists.opensips.org Message-ID: <1323972227724-7097996.p...@n2.nabble.com> Content-Type: text/plain; charset=us-ascii Hi all, I'm looking for a SIP/SIMPLE to XMPP gateway solution and my google searches have brought me here. I have an account with a voip provider that supports SMS via SIP/SIMPLE MESSAGE (http://tools.ietf.org/html/rfc3428). Unfortunately, I'm stuck using an old Blackberry which doesn't have any good SIP apps, the only one that I've found that supports RFC3428 doesn't work properly. I'm hoping that I can setup OpenSIPS to receive the SIP/SIMPLE messages from my voip provider and forward them to an XMPP server like openfire and be able to receive/respond to these messages via an XMPP client. Would this be possible with OpenSIPS? Also, would it be possible to have each message come from a unique XMPP user so that responses can be tracked to the proper source? Thanks. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-SIMPLE-to-XMPP-Gateway-for-SMS-tp7097996p7097996.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. -- Message: 5 Date: Thu, 15 Dec 2011 20:20:38 + (GMT) From: Logan Subject: Re: [OpenSIPS-Users] B2BUA Ripping/Truncating Callid To: OpenSIPS users mailling list Message-ID: Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Out of curiosity, based on the feedback in this bug; is this something that's being fixed? I notice this bug was for 1.6.4, but my experience is in 1.7.1. so I want to make sure if this was fixed, I report a new bug for 1.7.1 Hi Bogdan, This bug fix requires further work in tm module, in local_route processing, so as to update the shortcuts in tm when lumps are applied for headers also. The fix that was committed la
[OpenSIPS-Users] registration error
Hi, I'm new to OpenSIPS, set
up only load_balance
and REGISTER
but I have a problem with the registration.
returns error some users "Please
send new Register with
auth info 1000"
i use mysql for register users
if (is_method("REGISTER")) {
if (!www_authorize("", "subscriber")) {
if ($retcode < 0) {
switch ($retcode) {
case - 4:
xlog("L_INFO", "Please send new Register with
auth info $fU");
www_challenge("", "0");
exit;
case - 2:
xlog("L_INFO", "Wrong password for user $fU");
sl_send_reply("401", "Wrong password");
break;
case - 1:
xlog("L_INFO", "User doesnt exist $fU");
sl_send_reply("401", "User doesnt exist");
break;
}
}
}
if (!save("location")) {
sl_send_reply("408", "Requeste timeout");
exit;
}
exit;
}
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