Re: [OpenSIPS-Users] How to remove/update dialog
How to remove dialog from table dialog ? OR is there any method to update the
timeout value in dialog without calling create_dialog(B) ?
RegardsHamid R. Hashmi
From: hamid2kv...@hotmail.com
To: users@lists.opensips.org
Date: Tue, 10 Mar 2015 17:43:15 +0500
Subject: [OpenSIPS-Users] How to remove/update dialog
route[sip]{...t_on_failure(1);$DLG_timeout = 120;
create_dialog(B); t_relay();}
failure_route[1]{... if(t_check_status(some reasson)){
route(pstn); }...}
route[pstn]{...t_on_failure(2);$DLG_timeout = 60;
create_dialog(B);t_relay();}
How to remove previous dialog from table dialog ? OR is there any method to
update the timeout value in dialog without calling create_dialog(B) ?
RegardsHamid R. Hashmi
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[OpenSIPS-Users] [Release] OpenSIPS 2.1 Release Candidate is out !
Hi all, I'm really excited to announce that after almost one year of hard work, the OpenSIPS 2.1 RC is now available for download. http://opensips.org/pub/opensips/latest/src/ The 2.1 version is a major step in OpenSIPS evolution, encapsulating major redesign (protocols, timers, reactors, async support) and valuable additions (websockets, Fraud Detection, SIP Compression, Async DB queries, Topology Hiding and more). http://www.opensips.org/About/Version-2-1-0-Notes OpenSIPS 2.1 is the first in line benefiting of the New Design (NG) brainstorming and work - where many radical and innovative concepts have been introduced, to put OpenSIPS on the top of the SIP server engines - in terms of efficiency, scalability, features set and flexibility. OpenSIPS 2.1 is the result of the whole community effort - there were many people involved in the design, in implementation, testing and reporting - and we want to thank you all for making this release possible: https://github.com/OpenSIPS/opensips/blob/2.1/CREDITS Version 2.1 is a release candidate - the stable GA version is to be release in the next 1.5 month, after more in depth testing . In the mean while, enjoy the fantastic ride with OpenSIPS 2.1 :) Best regards, -- 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
Re: [OpenSIPS-Users] ERROR:dialog:dlg_validate_dialog:
I know you guys are super busy in OpenSIPS 2.1 release, but any suggestion
on above issue?
On Wed, Mar 18, 2015 at 12:17 AM, Satish Patel satish@gmail.com wrote:
I am getting following error in log, I can understand my contact: and
Route: values mismatching here. why it is happening? is there a way to get
rid on this error?
Following is scenario. Only getting error in BYE message.
[UA][OpenSIP]---[Freeswitch]-[Opensip]-[SIP
Provide]
ERROR:dialog:dlg_validate_dialog: failed to validate remote contact:
dlg=[sip:16463737221@188.178.235.222:5061;transport=udp] ,
req=[sip:188.178.235.222;lr;ftag=840e2e35;did=1f4.ca6a6956]
I am using fix_route_dialog() in loose_route()
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog()) {
if ($DLG_status!=NULL !validate_dialog() ) {
xlog( in-dialog bogus request \n);
fix_route_dialog();
}
xlog(L_INFO, Loose route failed on
$hdr(route)\n);
if (is_method(BYE)) {
#setflag(ACC_DO); # do accounting ...
#setflag(ACC_FAILED); # ... even if the
transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless,
do RR for
# re-INVITEs alos, as some buggy clients
do change route set
# during the dialog.
record_route();
}
if (check_route_param(nat=yes))
setflag(NAT);
# route it out to whatever destination was set by
loose_route()
# in $du (destination URI).
route(relay);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful
ACK; must be an ACK after
# a 487 or e.g. 404 from upstream
server
xlog(non loose-route section\n);
t_relay();
exit;
} else {
# ACK without matching transaction
-
# ignore and discard
xlog(ACK without matching
transaction\n);
exit;
}
}
xlog(L_INFO, destination uri after loose_route:
$du\n);
sl_send_reply(404,Not here);
}
exit;
}
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Re: [OpenSIPS-Users] WebSocket Support in OpenSIPS 2.1
Congrats to the team! Great work! On 17 Mar 2015, at 20:00, Răzvan Crainea raz...@opensips.org wrote: Hello, all! Aaand, we're finally making it official: OpenSIPS 2.1 will have *WebSockets* support! Are you planning to use (or perhaps you're already using) WebRTC based SIP clients, but you are having hard time setting up the platform? Starting from now, it has never been simpler - based on your needs and feedback[1] we decided to implement a WebSocket server directly in OpenSIPS. And we're doing it now! Starting with the new OpenSIPS 2.1 release you will be able to plug your web-based SIP clients directly in your OpenSIPS server using the new WebSocket transport protocol[2]. We've also setup a short tutorial[3] for you to integrate this feature in your platform easier. Many thanks to Eric Tamme (lirakis) for all his help with the tutorial as well as for the intensive tests. [1] http://www.opensips.org/Community/IRCmeeting20141029 [2] http://www.opensips.org/html/docs/modules/2.1.x/proto_ws [3] http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1 Best regards, -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Saúl Ibarra Corretgé AG Projects signature.asc Description: Message signed with OpenPGP using GPGMail ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Announcing rtpproxy v2.0.0
Nice reduction in poll overhead. Looking forward to trying this out. Any thoughts on adding epoll support for linux? It can tend to reduce overhead quite a bit with many sockets. On Mon, Mar 16, 2015 at 8:56 PM, John Mathew john.mat...@divoxmedia.com wrote: Yes On Tuesday, 17 March 2015, Zheng Frank zhengyumingap...@gmail.com wrote: Do you mean ROHC ? 2015-03-14 12:39 GMT+08:00 Maxim Sobolev sobo...@sippysoft.com: Do you have any particular RFC in mind? On Mar 12, 2015 10:28 AM, John Mathew john.mat...@divoxmedia.com wrote: Hi, Maxim, Is there any plans for rtp header compression in future. I can't see anything in the change log for 2.0.0 On Tuesday, 10 March 2015, Maxim Sobolev sobo...@sippysoft.com wrote: Hi All, I'm happy to announce that we have released rtpproxy v2.0.0. You can review the release notes here: https://github.com/sippy/rtpproxy/releases/tag/v2.0.0 -sobomax -- Sent from iPhone 6 -- You received this message because you are subscribed to the Google Groups rtpproxy group. To unsubscribe from this group and stop receiving emails from it, send an email to rtpproxy+unsubscr...@googlegroups.com. To post to this group, send email to rtppr...@googlegroups.com. To view this discussion on the web visit https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpUhu9fpwmqpRFiaXsQo9_%2B6M8MFtJ2sKi4kd5sr%3Ds%2BR5Q%40mail.gmail.com https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpUhu9fpwmqpRFiaXsQo9_%2B6M8MFtJ2sKi4kd5sr%3Ds%2BR5Q%40mail.gmail.com?utm_medium=emailutm_source=footer . For more options, visit https://groups.google.com/d/optout. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-us...@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Sent from iPhone 6 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] error 483
Hi; Can someone tell me why or where I may find the info I need; I'm able to register external remote phones (hard phones), but the internal phone (soft phones) within the same network as the OpenSIPS test server report error 483. Thanks!! Carlos ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dispatcher user specific route question
I have two Freeswitch in dispatcher table: +---+-+--+ | setid | destination | description | +---+-+--+ | 1 | sip:fs1.example.com | Freeswitch-1 | | 2 | sip:fs2.example.com | Freeswitch-2 | +---+-+--+ I have created zone column in subscriber table. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dispatcher user specific route question - 2.1
I have add extra zone column in subscriber table, +--+-+ | username | zone | +--+-+ |1001 |1| |1002 |2| +--+-+ In dispatcher table I have following two Freeswitch in two groups. +---+-++ | setid | destination | description| +---+--+---+ | 1 | sip:fs1.example.com | Freeswitch-1 | | 2 | sip:fs2.example.com | Freeswitch-2 | +---+--+---+ in opensips.cfg script i am query subscriber table base on incoming username and storing zone in avp(zone) variable, and calling same variable in following code if ( !ds_select_dst($avp(zone), 4, FM10)) Question: now either user belongs to zone 1 or 2, so it is *NOT* going to do load-balancing between two. But if I want to allow some user to do load-balancing then how it will be possible in above scenario? Can i set setid on fly so i can pass request along with user request and set same group for both switch and user call load-balance on both switch? Any other idea? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
