[SR-Users] Presence data to users behind NAT
Hello, Im running kamailio 4.2.3 and I'm trying to build a system where I could have the following features: SIP proxy, RTP proxy, Presence XCAP service for IM messaging. I have a problem with the Presence service with users that are located behind nat. When Kamailio sends presence info to a watcher with sip NOTIFY message, the destaination ip for the pacage seems to be an un-routable (=private) ip, and so the watcher never receives the NOTIFY message making the watchers sip client unaware of the other users presence. Any suggestions how to make Kamailio handle the watchers behind nat so that the NOTIFY messages will get delivered succesfully? I tried the scenario with a client with public ip and the presence works fine. my kamailio.cfg: http://tny.cz/7d85773c -- Olli ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio with websocket as transport
Thanks Daniel. Is there any example kamailio.cfg file with websocket enabled. I downloaded one from https://gist.github.com/jesusprubio/4066845/download# , it shows around 74 config errors. Thanks Austin On Fri, Mar 20, 2015 at 1:27 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 20/03/15 01:48, Austin Einter wrote: Dear All I have a sip user agent. Today it works fine with Kamailio using tcp as tranasport. Now I am planning to support websocket in my sip ua. My question is does Kamailio support websocket transport. If so how can I configure it or while building kamailio do I need to make some flags on. I am using latest Kamailio 4.2.3 (version: kamailio 4.2.3 (x86_64/linux)). kamailio supports websocket since version 4.0. Look at the readme for websocket module to get started with it: - http://kamailio.org/docs/modules/stable/modules/websocket.html If you need to decrypt/encrypt SRTP, then see rtpengine module and rtpengine application. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] remove_hf_value
Hello Daniel,
I end up insert routes like this. And right now on client side I see only
public ip route and private side only private ip routes.
route[ROUTE_HDR_ORG] {
# Getting record routes
if(is_present_hf(Route)) {
xlog(L_INFO, [$rm], Inbound connection from $si\n);
if(is_method(NOTIFY)) {
if(is_ip_rfc1918($si)) {
xlog(L_INFO, [$rm], Present Route: $(route_uri{uri.host})\n);
remove_hf(Route);
insert_hf(Record-Route: sip + $sel(cfg_get.kamailio.bindpublicip) + :
$sel(cfg_get.kamailio.bindpublicport);transport=tcp;lr=on;ftag=$ft;nat=yes\r\n);
$fs = tcp + : + $sel(cfg_get.kamailio.bindpublicip) + : +
$sel(cfg_get.kamailio.bindpublicport);
} else {
xlog(L_INFO, [$rm], Present Route: $(route_uri{uri.host})\n);
remove_hf(Route);
insert_hf(Record-Route: sip + : $sel(cfg_get.kamailio.bindip) + : +
$sel(cfg_get.kamailio.bindport);lr=on;ftag=$ft;nat=yes\r\n);
$fs = udp + : + $sel(cfg_get.kamailio.bindip) + : +
$sel(cfg_get.kamailio.bindport);
}
}
}
}
And for initial SUBSCRIBE INVITE
if (is_method(INVITE|SUBSCRIBE)) {
remove_hf(Route);
insert_hf(Record-Route: sip + : + $sel(cfg_get.kamailio.bindip) + : +
$sel(cfg_get.kamailio.bindport);lr=on;ftag=$ft;nat=yes\r\n);
#record_route();
}
Slava.
From: Daniel-Constantin Mierla mico...@gmail.com
To: Slava Bendersky volga...@networklab.ca, sr-users
sr-users@lists.sip-router.org
Sent: Wednesday, March 18, 2015 4:15:28 AM
Subject: Re: [SR-Users] remove_hf_value
Hello,
but the INVITE request goes out with 2 Record Route headers, if you get them in
the 200ok.
Not sure how it is going to work overall, you will have to manually set local
sockets for requests within dialog, because you are doing also trasport layer
gatewaying (tcp-udp).
Anyhow, as a solution, maybe you should just not be using record_route()
function and just add the header with insert_hf() for the invite, remove the
Record-Route header for 200ok with remove_hf() and insert a new one again with
insert_hf()
If you know the local address (IP/port/transport) you want to put in the
Record-Route, you can build the header as:
insert_hf(Record-Route:
sip:_LOCAL_IP_:_LOCAL_PORT_;transport=_LOCAL_TRANSPORT_;lr=on;ftag=$ft;nat=yes
\r\n);
Requests within the dialog can be handled with loose_route(), but again, you
have to do more in config for selecting the local socket to send to next hop.
Cheers,
Daniel
On 16/03/15 13:35, Slava Bendersky wrote:
Hello Daniel,
I want do clean up on routes from first SUBSCRIBE, that when traffic goes to
public side of kamailio will be only public route and when traffic is internal
the Record-Route will stay internal only. In SDP I see like this.
I tired double routes set 0 in rr, but didn't clean enough.
--- Transmitting (NAT) to 10.18.130.46:5060 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060
Via: SIP/2.0/TCP
192.168.88.252:5062;received=client_pub_ipbranch=z9hG4bK0bbe1f7d27257bba9;rport=5062
Record-Route: sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes On public
side remote this Record-Route
Record-Route:
sip:kamailio_pub_ip:5084;transport=tcp;r2=on;lr=on;ftag=a185d974ec;nat=yes On
private side remote this Record-Route
From: Slava Bendersky sips:10...@networklab.ca ;tag=a185d974ec
To: sips:10...@networklab.ca ;tag=as00757d3e
Call-ID: b08adb1ad1804a83
CSeq: 236711034 SUBSCRIBE
Server: FPBX-2.11.0(11.15.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: sip:10101@10.18.130.50:5060 ;expires=3600
Content-Length: 0
Slava.
From: Daniel-Constantin Mierla mico...@gmail.com
To: sr-users sr-users@lists.sip-router.org
Sent: Monday, March 16, 2015 7:01:54 AM
Subject: Re: [SR-Users] remove_hf_value
Hello,
can you give an example of incoming header and how you want to be after
modification?
It is not clear for me what end result you want/expect.
Cheers,
Daniel
On 14/03/15 06:41, Slava Bendersky wrote:
BQ_BEGIN
Hello Everyone,
Here updated version. Trying remove Record-Route after the coma.
My log shows
Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: script:
[SUBSCRIBE], Inbound connection from Client Public IP
Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: ERROR: script: null
Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: script:
[SUBSCRIBE], Route for removal: 'null'
route snippet.
route[ROUTE_HDR_ORG] {
# Getting record routes
if(is_present_hf(Route)) {
xlog(L_INFO, [$rm], Inbound connection from $si\n);
if(is_method(SUBSCRIBE)) {
if(is_ip_rfc1918($si)) {
xlog( $sel(@hf_value.Record_Route[-1]) );
xlog(L_INFO, [$rm], Route for removal:
'$sel(@hf_value.Record_Route[-1])'\n);
remove_hf_value(Record-Route[-1]);
} else {
xlog( $sel(@hf_value.Record_Route[1]) );
xlog(L_INFO, [$rm], Route for removal:
'$sel(@hf_value.Record_Route[1])'\n);
Re: [SR-Users] kamailio asterisk NOTIFY
Hello Daniel, I still can't understand clearly why when asterisk send NOTIFY to client through kamailio it coming back with with 408. I I tried different case event change Contact:, but no luck can't I see what the issue all routes in first SUBSCRIBE looks ok . Any though or help will be really appreciate. This what I see on asterisk debug Reliably Transmitting (NAT) to 10.18.130.46:5060: NOTIFY sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK5dd3f266 Max-Forwards: 70 Route: sip:10.18.130.46:5060;lr=on;ftag=3b27a2a409;nat=yes ---Kamailio route From: canlpbx02 sip:canlpbx02@10.18.130.51;tag=as56402f92 To: sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2;tag=3b27a2a409 Contact: sip:canlpbx02@10.18.130.51:5060 Call-ID: b1edde040d695988 CSeq: 102 NOTIFY User-Agent: FPBX-12.0.43(12.8.1) Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 88 Messages-Waiting: yes Message-Account: sip:*97@10.18.130.51 Voice-Message: 5/0 (0/0) --- --- SIP read from UDP:10.18.130.46:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.51:5060;rport=5060;branch=z9hG4bK5dd3f266 From: canlpbx02 sip:canlpbx02@10.18.130.51;tag=as56402f92 To: sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2;tag=3b27a2a409 Call-ID: b1edde040d695988 CSeq: 102 NOTIFY Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE Server: Media5-fone/4.1.6.3283 Android/5.1 Supported: eventlist, replaces, timer Content-Length: 0 - --- (10 headers 0 lines) --- --- SIP read from UDP:10.18.130.46:5060 --- SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6386387f;rport=5060 From: canlpbx02 sip:canlpbx02@10.18.130.51;tag=as00936a6d To: sip:102@192.168.88.243:5066;alias=client_pub_ip~5066~2;tag=a3aea18741 Call-ID: 5a66b87a618d1042 CSeq: 103 NOTIFY Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0 - Slava. From: Daniel-Constantin Mierla mico...@gmail.com To: Slava Bendersky volga...@networklab.ca, sr-users sr-users@lists.sip-router.org Sent: Friday, March 13, 2015 1:28:33 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, that function might be in textopsx module. Cheers, Daniel On 13/03/15 14:25, Slava Bendersky wrote: Hello Daniel, Where possible find updated version of options for textops. [sr_module.c:701]: find_mod_export_record(): find_export_record: remove_hf_value not found Specially Record-Route manipulation. Slava. From: Slava Bendersky volga...@networklab.ca To: mico...@gmail.com Cc: sr-users sr-users@lists.sip-router.org Sent: Friday, March 13, 2015 8:54:26 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, Here paste for first SUBSCRIBE. http://fpaste.org/197545/25118514/ Slava. From: Daniel-Constantin Mierla mico...@gmail.com To: Slava Bendersky volga...@networklab.ca Cc: sr-users sr-users@lists.sip-router.org Sent: Thursday, March 12, 2015 5:09:17 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, can you get the sip traffic from the network (from the first subscribe of this presence dialog)? It will help to see the Route set. Cheers, Daniel On 12/03/15 00:36, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here debug of the notify which failing. Look like transaction miss match http://fpaste.org/196896/61168591/ Slava. From: Daniel-Constantin Mierla mico...@gmail.com To: Slava Bendersky volga...@networklab.ca , sr-users sr-users@lists.sip-router.org Sent: Wednesday, March 11, 2015 5:44:05 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, it doesn't matter if there is a mix of transports. If the UA is mirroring the Record-Route headers in the response to the NOTIFY, it should work fine. Cheers, Daniel On 11/03/15 21:40, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Is this matter that UDP and TCP transport in mix ? Slava. From: Slava Bendersky volga...@networklab.ca To: mico...@gmail.com Cc: sr-users sr-users@lists.sip-router.org Sent: Wednesday, March 11, 2015 4:21:39 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, I have record route in WITHINDLG for NOTIFY, but it not helped behaviour is the same it loosing route for NOTIFY. Slava. From: Daniel-Constantin Mierla mico...@gmail.com To: Slava Bendersky volga...@networklab.ca , sr-users sr-users@lists.sip-router.org Sent: Wednesday, March 11, 2015 2:14:02 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, you have to do record_route() for all NOTIFY requests -- there are new specs requiring that (RFC 6665) and perhaps asterisk is following them. If you look in the default kamailio.cfg for v4.2, inside route[WITHINDLG], then you will see how it is done. Cheers, Daniel On 11/03/15 14:22, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here
[SR-Users] Websocket kamailio.cfg doubt
The README of websocket suggests to modify kamailio.cfg as below
*# Handle requests within SIP dialogsroute[WITHINDLG] {if
(has_totag()) {# sequential request withing a dialog
should# take the path determined by
record-routingif (loose_route()) {
if ($du == ) {if (!handle_ruri_alias())
{xlog(L_ERR, Bad alias
$ru\n);sl_send_reply(400, Bad
Request);
exit;}
}route(RELAY);} else
{if ( is_method(ACK) ) {...*
But the original script was
*# Handle requests within SIP dialogsroute[WITHINDLG] {if
(!has_totag()) return;# sequential request withing a dialog
should# take the path determined by record-routingif
(loose_route()) {route(DLGURI);if
(is_method(BYE)) {setflag(FLT_ACC); # do
accounting ...setflag(FLT_ACCFAILED); # ... even if
the transaction fails}else if (
is_method(ACK) ) {# ACK is forwarded
statelessyroute(NATMANAGE);
}else if ( is_method(NOTIFY) ) {#
Add Record-Route for in-dialog NOTIFY as per RFC
6665.record_route();
}route(RELAY);exit;}if
(is_method(SUBSCRIBE) uri == myself) {# in-dialog
subscribe requestsroute(PRESENCE);
exit;}if ( is_method(ACK) ) {if (
t_check_trans() ) {# no loose-route, but stateful
ACK;# must be an ACK after a
487# or e.g. 404 from upstream
serverroute(RELAY);
exit;} else {# ACK without matching
transaction ... ignore and discard
exit;}}sl_send_reply(404,Not
here);exit;}*
Now I have to include the websocket README suggested script in original
script. I am not finding logical way where shall I include websocket part.
Can somebody please help me.
Thanks
Austin
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Re: [SR-Users] Siremis error Call to a member function getQuoted()
I think I found the root cause. Siremis database has a table called group. That is a reserved word in mysql that cannot be used as a table name. http://dev.mysql.com/doc/refman/5.5/en/reserved-words.html Do the following to prove it. mysql USE siremis; SELECT * FROM group; error RENAME TABLE group TO group1; SELECT * FROM group1; no error On 3/21/2015 4:39 PM, canuck15 wrote: When you add/update/view Administrator usernames To reproduce Administration User Management Click on any existing username. The browser goes blank. Error in apache logs is: PHP Fatal error: Call to a member function getQuoted() on a non-object in /var/www/html/openbiz/bin/data/BizDataSql.php on line 237 This is on the latest Siremis v4.2.0 downloaded today I am using CentOS v6.6, php v5.3.3 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
