[OpenSIPS-Users] Jitsi sent REGISTER, Opensips received nothing

[jacky]

I have a test with one Jitsi using Opensips on the Internet
Wireshark showed me that Jitsi sent several "REGISTER" packages, in the same
time I used the command "tcpdump" to listen on the Opensips Server , but got
nothing.
Anyway, on the server, opensips, rtpproxy, media-dispatcher, media-relay are
running!
what happened to opensips server? why it won't response to distanced
request?

I appreciate your opinion, thanks a lot!

Best regards




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[OpenSIPS-Users] Dispatcher user specific route question - 2.1

[Satish Patel]

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?
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[OpenSIPS-Users] Dispatcher user specific route question

[Satish Patel]

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.
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[OpenSIPS-Users] error 483

[Carlos Cruz]

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

 

 

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Re: [OpenSIPS-Users] Announcing rtpproxy v2.0.0

[Ryan Bullock]

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 
wrote:

> Yes
>
> On Tuesday, 17 March 2015, Zheng Frank  wrote:
>
>> Do you mean ROHC ?
>>
>> 2015-03-14 12:39 GMT+08:00 Maxim Sobolev :
>>
>>> Do you have any particular RFC in mind?
>>> On Mar 12, 2015 10:28 AM, "John Mathew" 
>>> 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  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
>
>

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>>>
>>> ___
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>>> sr-us...@lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>
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Re: [OpenSIPS-Users] ERROR:dialog:dlg_validate_dialog:

[Satish Patel]

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  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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[OpenSIPS-Users] [Release] OpenSIPS 2.1 Release Candidate is out !

[Bogdan-Andrei Iancu]

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


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Re: [OpenSIPS-Users] How to remove/update dialog

[Hamid Hashmi]

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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Re: [OpenSIPS-Users] WebSocket Support in OpenSIPS 2.1

[Saúl Ibarra Corretgé]

Congrats to the team! Great work!

On 17 Mar 2015, at 20:00, Răzvan Crainea  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
> 
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--
Saúl Ibarra Corretgé
AG Projects





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