Hello All,
Please can anyone give some information regarding dispatcher.list file.
I am using LAN network and trying to setup a kamailio-ims server.
What is the info I need to configure in dispatcher.list ?
I see error while running pcscf module.
Thanks
Hello,
in preparation for releasing v4.2.0 on Thursday this week, I spent some
time today to publish the documentation for v4.2.x series at the usual
places on kamailio.org site.
I am writing here to ask everyone having some spare time to check
through usual places they used in the past to read documentation on
kamailio.org and see if something is missing for 4.2.x or something is
wrong (e.g., broken or wrong links). Starting points can be:
- http://www.kamailio.org/w/documentation/
- http://www.kamailio.org/wiki/
Report any finding back to mailing list.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
just to complete the thread, the dependency problem has been fixed and
upgrade from 3.3.0.0+0~mr3.5.0.0 to 3.3.0.0+0~mr3.6.0.0 succeeded
without issues.
-- juha
Hi guys,
I've setup my kamailio more or less with all my needs, thanks to all your
helps :)
Now I've a question regarding presence.
My kamailio manage all sip clients that are presents in my home. When I
receive call all my phone rings simultaneously and so on...
I would like that I can manage presence status of my home in order that...if
some of my phones is "on call" the status of my "group" is "on call". It's
something like that my "home" presence status is the AND/OR of all my phones
presence status...
Is there any way to make this? I think that I've to use PUA module, but the
problem is that multiple calls are possible so is there any way to
understand from script number of ongoing calls?
-----
Marino Mileti
--
View this message in context: http://sip-router.1086192.n5.nabble.com/Group-User-Presence-tp131478.html
Sent from the Users mailing list archive at Nabble.com.
Following this asipto guide:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
I've found that no auth is required to create a communication: Invites are
sent regardless of a previously done auth with the kam's auth_db
I've used this sip checker:
http://www.sinologic.net/proyectos/asterisk/checkSecurity/
Result:
Uh oh! you allow external calls...
Configure better your sip configuration to avoid this calls
SIP/2.0 100 trying -- your call is important to us
What is missing in the config file explained in the guide?
Manuel
Greetings,
I'm running into a slow memory leak on my kamailio 4.0.4 SIP proxies. I'm observing a steady increase in the memory consumption until there is no more left. Kamailio then starts repeating this in the logs:
ERROR: dispatcher [dispatch.c:279]: add_dest2list(): no more memory.
What would be the best way to debug this kind of a memory leak? The proxy does not handle any registrations but does route a fair amount of calls.
Thank you in advance,
Tim Heenan
Engineer I - VoIP Wholesale | Windstream
timothy.heenan(a)windstream.com<mailto:email@windstream.com> | windstreambusiness.com<http://www.windstreambusiness.com/>
o: 847.348.1338 | f: 847.963.0116
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Hello there,
I upgraded my kamailio version from 4.0.6 to 4.1.6 and now when i try start
kamailio it breaks down on htable module initialization, as you can see on
logs bellow:
0(1273) DEBUG: <core> [sr_module.c:966]: init_mod(): DEBUG: init_mod:
presence_profile
0(1273) DEBUG: <core> [sr_module.c:701]: find_mod_export_record():
find_export_record: found <bind_presence> in module presence
[/usr/local/lib64/kamailio/modules/presence.so]
0(1273) DEBUG: presence [event_list.c:351]: search_event(): start event=
[ua-profile/7]
0(1273) DEBUG: presence [event_list.c:351]: search_event(): start event=
[ua-profile.winfo/0]
0(1273) DEBUG: presence [event_list.c:259]: add_event(): successfully
added event: ua-profile - len= 10
0(1273) DEBUG: <core> [sr_module.c:966]: init_mod(): DEBUG: init_mod:
htable
0(1273) ERROR: htable [ht_api.c:282]: ht_init_tables(): no more shm for
[dlg]
0(1273) ERROR: <core> [sr_module.c:970]: init_mod(): init_mod(): Error
while initializing module htable
(/usr/local/lib64/kamailio/modules/htable.so)
ERROR: error while initializing modules
0(1273) : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing
already freed pointer (0x7fae605547c0), called from htable: ht_api.c:
ht_destroy(343), first free htable: ht_api.c: ht_destroy(343) - aborting
0(1271) ERROR: <core> [daemonize.c:307]: daemonize(): Main process exited
before writing to pipe
The shared memory allocated for kamailio is 256MB and is the same that i
had for version 4.0.1, can this be a bug?
Thank you for your support
--
Cumprimentos
José Seabra
Hi Andrew,
I just use git... so not sure on the rpm stuff. However I believe 4.2 is
to be released as stable branch tomorrow (see other recent thread by
Daniel) so the packages probably aren't far behind.
The main reasons I see for using the 4.2 branch are:
1) its going to be stable release very soon
2) the options on rtpengine have been refactored ... makes sense to
develop on the new version so that you can move from development to
production without amending your kamailio.cfg
3) rtpengine seems to be getting quite alot of rapid progress at the
moment and gaining stability and features ... I'd rather be on the
latest version.
Having said all that the logic in the kamailio.cfg for version 4.1 is
the same and should work for you if you want to work from the current
stable packages.
On 13/10/14 11:10, andrew wrote:
> hi,
>
> thanks for your help.
>
> rtpengine module is used only in kamalio 4.2.x ------ development
> version. where to get its rpm packet?
>
> B.R.
>
> andrew
>
>
> At 2014-10-13 16:38:47, "Paul Smith" <paul.smith(a)claritytele.com> wrote:
>
> Hi Andrew
> There is a condition in the NATMANAGE route which tests whether or
> not to apply rtpproxy_manage() :
>
> if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
> return;
>
> One way to force this is to make sure that FLT_NATS is always set.
> That can be done in the NATDETECT route by moving
> setflag(FLT_NATS) outside of the if(nat_uac_test) condition and
> putting it just before the return statement, so the flag is always
> set.
>
> Another way would be to remove or comment out the if
> "!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return;" line
> from NATMANAGE so that the the rtp_manage() is always called.
>
> It might be worth your while upgrading to the 4.2 branch which has
> renamed the rtpproxy-ng module to rtpengine and made the options
> in the call to rtpengine_manage() more readable.
>
> Depending on why you always want to run the rtp proxy ... you
> might also want to set :
>
> rtpengine_manage(replace-session-sonnection replace-origin ICE=force-relay)
>
> to ensure that media always goes through your media proxy and no
> other connections are negotiated between the endpoints.
>
> On 13/10/14 08:36, andrew wrote:
>> hi,
>>
>> kamailio 4.1.5 is used with rtpengine, whose older counterpart is
>> rtpproxy-ng. I made some changes based on the default
>> configuration files(i.e. kamailio.cfg), so that rtpproxy-ng
>> module is enabled. Kamailio.cfg has been uploaded. In
>> kamailio.cfg, there is one route block related to rtpproxy, i.e.
>> route[NATMANAGE], where rtpproxy_manage("co") is called.
>> Sometimes, rtpproxy-ng can insert new ip:port candidates in SDP.
>> But in some case, rtpproxy-ng doesn't rewrite SDP, so rtpengine
>> doesn't relay rtp packets at all.
>>
>> How to setup kamailio.cfg, so that rtpproxy-ng moudle can always
>> rewritten new ip:port candidates, and rtpengine always relays rtp
>> packets?
>>
>> Looking foreward to your reply. Thanks!
>>
>> B.R.
>>
>> Andrew
>>
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users(a)lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
Thanks for the feedback.
On 14/10/14 02:13, Alberto Llamas wrote:
> Hello Paul,
>
> That was amazing. Thanks a lot !
>
> On Mon, Oct 13, 2014 at 3:42 AM, Paul Smith
> <paul.smith(a)claritytele.com <mailto:paul.smith@claritytele.com>> wrote:
>
> Hi Alberto
> The magic you are looking for is in the "domain" module. The
> example configuration files in the distribution have it set up and
> ready to use if you add the line "#!define WITH_MULTIDOMAIN"
>
> The important bit for your scenario is in the example config:
>
> # ----- domain params -----
> #!ifdef WITH_MULTIDOMAIN
> modparam("domain", "db_url", DBURL)
> # register callback to match myself condition with domains list
> modparam("domain", "register_myself", 1)
> #!endif
>
> That section tells the domain module to use the database, and to
> include the domains loaded from there when matching the special
> keyword "myself". The example config files also set up other
> modules (like user location module) to be domain aware as well.
>
> In this setup you do not need to restart kamailio when you add or
> remove domains.... but you do need to tell it to reload the
> domains from database as Kamailio holds a cached copy of the list
> in memory.... so when you make a database change then do "kamctl
> domain reload". You can also run "kamctl domain show" to see what
> domains are currently listed in memory.
>
>
>
> On 12/10/14 21:38, Alberto Llamas wrote:
>>
>> Hello kamailio gurus,
>>
>> Thanks if somebody can help me answering nex question:
>>
>> I have a kamailio server as registrar server with asterisk
>> behind. For register porpouse the UA are using subdomain like:
>>
>> 100(a)test.mydomain.com <mailto:100@test.mydomain.com>
>> 200(a)test2.mydomain.com <mailto:200@test2.mydomain.com>
>>
>> I have to add each alias in the kamailio.cfg file an restart the
>> sevice.
>>
>> Alias="test.mydomain.com <http://test.mydomain.com>"
>> Alias="test2.mydomain.com <http://test2.mydomain.com>"
>>
>> Is there some option to add it in the database? Then i don't have
>> to restart the server each time i have to add/edit/delete each
>> subdomain.
>>
>> Thanks a lot,
>>
>> Albert
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>
> --
> Alberto Llamas
> TelecommunicationsEngineer
> Digium Certified Asterisk Professional (dCap)
> Linux Administrator
>