[SR-Users] RFC-5766 / RFC-6062 compliant TURN clients?
I know this might not be totally on-topic for the list, but I figured the list participants might have experience in this domain, so I'm lobbing it out there... I've got some situations where I'd like to connect users who are oftentimes sitting on very locked-down networks (ie, corporate networks) to some SIP-based VoIP infrastructure, but the idea of opening random UDP ports on the networks in question is a no-go. As a result, I'm experimenting with TURN over TCP or TLS as a solution to enable connectivity... however, I can't find any clients that implement the latest specs (X-Lite 4 appears to speak an older version of the TURN protocol; QJsip doesn't work for me; etc. etc.). Do folks here have any suggestions for TURN-compatible clients? I care mostly about PC for the time being (ie, just a proof of concept) but a PC/Mac/Linux solution would be great too. Thanks for any insights, --rafal ___ 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] RFC-5766 / RFC-6062 compliant TURN clients?
Am 22.08.2011 21:52, schrieb Rafal Boni: I know this might not be totally on-topic for the list, but I figured the list participants might have experience in this domain, so I'm lobbing it out there... I've got some situations where I'd like to connect users who are oftentimes sitting on very locked-down networks (ie, corporate networks) to some SIP-based VoIP infrastructure, but the idea of opening random UDP ports on the networks in question is a no-go. As a result, I'm experimenting with TURN over TCP or TLS as a solution to enable connectivity... however, I can't find any clients that implement the latest specs (X-Lite 4 appears to speak an older version of the TURN protocol; QJsip doesn't work for me; etc. etc.). ICE/TURN is not activated in QjSimple. Guess I should do that. Is TURN over TCP/TLS a standard? regards Klaus ___ 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
[SR-Users] max branches
Hi all Is there any configuration option in kamailio 3.1 to set the max number of branches? I have a system that needs to run over 30 times to failure_route and call append_brach() from there. At 11-12 iteractions I get the following errors: ERROR: tm [t_fwd.c:651]: ERROR: add_uac: maximum number of branches exceeded ERROR: tm [t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches ERROR: tm [tm.c:1368]: ERROR: w_t_relay_to: t_relay_to failed ERROR: tm [t_reply.c:965]: ERROR: run_failure_handlers: Error in run_top_route cheers, Jon ___ 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
[SR-Users] Development branch frozen for v3.2.0
Hello, since several hours ago, the development of new features in master branch has been frozen, to allow proper testing for release of version 3.2.0. That means no new features should be committed to GIT master branch. Just before the release (expected in 1.5months) we will create a dedicate branch for 3.2.x series and master branch will be again open for new stuff. Meanwhile, developers can use personal branches to push code in the public space. GIT master branch will be used in this period for: - bug fixing - hammering of the new features in 3.2.0 - documentation improvements - merging of duplicated modules There are two good news for 3.2.0: - the number of the new features is very impressive. Besides the old squad, we gained many new developers lately, coming with interesting contributions, therefore I am sure you'll have lot of fun testing them - the core components (transport layers, asynchronous processing, timers, locking memory managers, a.s.o.) were barely touched since 3.0, so this is the 3rd release in a row using the new scalable core framework. That shows the maturity and there will be no much to focus on them If you need assistance while trying some new features, feel free to email to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ 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
[SR-Users] Testing upcoming version 3.2.0
Hello, for those looking to test v3.2.0 and need some instructions, for Kamailio flavour is a wiki page showing step-by-step installation: * http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-devel-from-git Practically 3.2.0 is the current devel branch. Testing SER flavour the difference is just using FLAVOUR=ser when compiling the application. Developers as well as users are encouraged to add details about the new additions in v3.2.0 to wiki page: * http://sip-router.org/wiki/features/new-in-devel Also, please add instructions for migrating from 3.1.x to 3.2.0 to the wiki page: * http://sip-router.org/wiki/install/3.1.x-to-3.2.x Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ 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] max branches
Hola Jon, It can be changed at config.h: /* maximum number of branches per transaction */ #define MAX_BRANCHES12 But not sure if there is some variable in the configuration script. Saludos JesusR. El 23/08/2011, a las 11:18, Jon Bonilla (Manwe) escribió: Hi all Is there any configuration option in kamailio 3.1 to set the max number of branches? I have a system that needs to run over 30 times to failure_route and call append_brach() from there. At 11-12 iteractions I get the following errors: ERROR: tm [t_fwd.c:651]: ERROR: add_uac: maximum number of branches exceeded ERROR: tm [t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches ERROR: tm [tm.c:1368]: ERROR: w_t_relay_to: t_relay_to failed ERROR: tm [t_reply.c:965]: ERROR: run_failure_handlers: Error in run_top_route cheers, Jon ___ 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 Saludos JesusR. Jesus Rodriguez VozTelecom Sistemas, S.L. jes...@voztele.com http://www.voztele.com Tel. 902360305 - ___ 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
[SR-Users] uac_req: pseudovariables, avps
Hi all
I'm trying to generate a request using uac_req in a failure route when the call
has been canceled. (Kamailio 3.1.3)
Here's the code:
failure_route[FAILURE_ROUTE_LALA]
{
if t_is_canceled()
{
$uac_req(method) = INVITE;
$uac_req(ouri) = sip:127.0.0.1:5090;
$uac_req(ruri) = $ru;
$uac_req(furi) = $fu;
$uac_req(turi) = $tu;
$uac_req(hdrs) = X-PUSH-Type: mytype\r\nX-PUSH-CLI:
$avp(s:first_caller_cli)\r\nX-PUSH-DST: $rU\r\n;
uac_req_send(); route(ROUTE_STOP_RTPPROXY);
exit;
}
...
}
The problem here is that the headers X-PUSH-CLI and X-PUSH-DST have literal
'$avp(s:first_caller_cli)' and '$rU' values. Here's the INVITE I send:
U 2011/08/23 17:07:09.467666 127.0.0.1:5062 - 127.0.0.1:5090
INVITE sip:ngcpte...@domain.com SIP/2.0'
Via: SIP/2.0/UDP
127.0.0.1:5062;branch=z9hG4bK925.85ac80a2.0'
To: sip:ngcpte...@domain.com'
From: sip:sipwi...@domain.com;tag=cf3b49e60a035342e3af7df009437068-9f9a'
CSeq: 10 INVITE'
Call-ID: 25f2b3a34ed8488c-3417@127.0.0.1'
Content-Length: 0'
User-Agent: Sipwise NGCP Proxy 2.X'
X-PUSH-Type: mytype'
X-PUSH-CLI: $avp(s:first_caller_cli)'
X-PUSH-DST: $rU'
I also tried to put the headers in a $var and in a $avp and set the value like:
$var(cancel_push_request)=X-PUSH-Type: missed_call\r\nX-PUSH-CLI:
$avp(s:first_caller_cli)\r\nX-PUSH-DST: $rU\r\n;
$uac_req(hdrs) = $var(cancel_push_request);
If I xlog the values like this:
xlog(L_INFO, Comprobando valores '$avp(s:first_caller_cli)' --
'$rU');
The kamailio log show the correct value:
INFO: script: Comprobando valores '4312345' -- 'ngcptest1'
How could I use my stored values in the INVITE generation?
cheers,
Jon
___
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Re: [SR-Users] uac_req: pseudovariables, avps
On Tuesday 23 August 2011, Jon Bonilla wrote: Hi all I'm trying to generate a request using uac_req in a failure route when the call has been canceled. (Kamailio 3.1.3) Here's the code: $uac_req(hdrs) = X-PUSH-Type: mytype\r\nX-PUSH-CLI: $avp(s:first_caller_cli)\r\nX-PUSH-DST: $rU\r\n; Strings aren't evaluated for PV's. Use pv_printf or concatenate the different components with '+'. -- Greetings, Alex Hermann ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] WARNING: timer: add_timeout: 0 expire timer added
Hello, I get this warning message when a request is relayed over tcp: WARNING: core [timer_funcs.h:119]: WARNING: timer: add_timeout: 0 expire timer added The warning is displayed after send_route() has run. (relevant) settings: modparam(tm, fr_timer, 500) modparam(tm, fr_inv_timer, 8) modparam(tm, wt_timer, 2) Every other timer related setting is at its default. Just before t_relay() is called, fr timer is set with: t_set_fr($avp(ringtime) * 1000); where $avp(ringtime) = 80. An identical setup with udp does not show the warning. Where does this warning come from and how do i suppress it? -- Greetings, Alex Hermann ___ 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] uac_req: pseudovariables, avps
El Tue, 23 Aug 2011 17:32:41 +0200 Alex Hermann a...@speakup.nl escribió: On Tuesday 23 August 2011, Jon Bonilla wrote: Hi all I'm trying to generate a request using uac_req in a failure route when the call has been canceled. (Kamailio 3.1.3) Here's the code: $uac_req(hdrs) = X-PUSH-Type: mytype\r\nX-PUSH-CLI: $avp(s:first_caller_cli)\r\nX-PUSH-DST: $rU\r\n; Strings aren't evaluated for PV's. Use pv_printf or concatenate the different components with '+'. Thanks Alex. It has sense. Tried with '+' and worked perfectly. ___ 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
[SR-Users] WARNING: timer: add_timeout: 0 expire timer added
Alex Hermann writes: I get this warning message when a request is relayed over tcp: WARNING: core [timer_funcs.h:119]: WARNING: timer: add_timeout: 0 expire timer added which version of sr is that? have you tried with latest master version? -- juha ___ 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] WARNING: timer: add_timeout: 0 expire timer added
On Tuesday 23 August 2011 14:57:51 Juha Heinanen wrote: Alex Hermann writes: I get this warning message when a request is relayed over tcp: WARNING: core [timer_funcs.h:119]: WARNING: timer: add_timeout: 0 expire timer added which version of sr is that? have you tried with latest master version? It is the latest master version, Kamailio flavour. Btw, I have not seen this on a 3.0.1, SR flavour installation (simpler config). If the cause is not obvious to anyone, i'll try to reproduce with a trivial script. The issue is showing itself in a huge config with many, possibly interfering, modules loaded. -- Alex Hermann ___ 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] WARNING: timer: add_timeout: 0 expire timer added
Alex Hermann writes: which version of sr is that? have you tried with latest master version? It is the latest master version, Kamailio flavour. andrei fixed this warning a couple of days ago and it disappeared from my setups. if you have build your sip router from today's master version and still get the warning, then looks like there is another incarnation of the bug still to be fixed. -- juha ___ 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] WARNING: timer: add_timeout: 0 expire timer added
On Tuesday 23 August 2011 15:29:27 Juha Heinanen wrote: andrei fixed this warning a couple of days ago and it disappeared from my setups. if you have build your sip router from today's master version and still get the warning, then looks like there is another incarnation of the bug still to be fixed. Indeed, I missed that commit in the stream of commits and was still on an ancient version of saterday evening... Current master is ok. Thanks, -- Alex Hermann ___ 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] Float Comparison
Daniel, Sorry for taking so long to respond - I am still facing an issue with this (even in 3.1.4). I am loading the values from database. It looks as if the arithmetic operators are not functioning properly when the data is imported into kamailio from DB - I'm going to do some more research and give you some specifics. Sincerely, Brandon Armstead On Thu, Jan 20, 2011 at 2:13 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: On 1/19/11 7:50 AM, Klaus Darilion wrote: Am 18.01.2011 21:26, schrieb Brandon Armstead: Hello, Is there anything special that needs to be done for float comparison? For example: if([5.5 = 4.3]) ^^^ this format is no longer supported starting with 3.0, just skip the square brackets, now it is working like in C. or if(5.5 4.3) The conditional does not seem to be coming back as true like it should? I have no idea if floating point comparison is supported, but you could multiple the values (e.g. * 1) before comparison The pseudo-variables can hold integer or strings. Do you do comparison with static values or you load the values in some variables and then compare? Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.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
[SR-Users] SIP to SMS/HTTP
Hello list, I am not sure if this has been ask before but is there a way that Kamailio can sip received SIP method MESSAGES and then call a URL (Kannel URL). Thank you. Regards, Mark Anthony C. Delfin ___ 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
