Hello,
https://www.kamailio.org/docs/modules/devel/modules/dialog#dialog.p.dlg_fil… says for dialog.dlg_filter_mode (int)
Set dialog fitering mode, which can specify what dialogs are processed. Its value can be a combination (the sum) of following flags:
1 - do not send keepalives and do not execute timeout function if dialog is not local (if the associated bind address is not a local socket).
Default value is “0”.
As you can see, the definition of 1 is a double negation and there is no definition for 0. What means 0:
- do not send keepalives and do not execute timeout function if dialog is local
- send keepalives and execute timeout function if dialog is not local
- send keepalives and execute timeout function if dialog is local
Hello,
the documentation for tm.t_newtran() says „This is the only way a script can add a new transaction in an atomic way.“ Moreover tm.t_send_reply()
“creates the transaction if it does not exist (executing internally t_newtran()) and sends a stateful reply (executing internally t_reply())”. This
means that only t_newtrans() and t_send_reply() can create transactions in an atomic way.
The documentation of tm.t_relay() does not say whether the function creates a transaction. It also does not say what happens if t_reply() is not
called after t_newtrans().
But for tm.t_set_retr() and tm.t_set_retr() is written “If the transaction is already created (e.g called after t_relay() or in an onreply_route) …”
This suggests that tm.t_relay() does create atomically or not-atomically a transaction.
- What does it mean to create a transaction in a non-atomic way?
- Does tm.t_relay() create a transaction?
Greetings
Dilyan
Hello everybody,
I'm working on a kamailio instance as a basic registrar service and I thought to use db_postgres to connect a postgres service only allowing SSl connections.
I see that that module doesn't provide any parameters to select sslmode and I was wondering if this module actually supports TLS connections.
Does anybody know if this is supported and, if so, how to enable it? or should that support be added into db_postgres module code?
Thank you very much in advance.
BR,
Joel Centelles
RTC Lead Engineer | Care Communications R&D
joel_centellesmartin(a)baxter.com<mailto:joel_centellesmartin@baxter.com> | baxter.com<https://baxter.com/>
[cid:9ee6bcbd-9c30-45ec-bb43-8f439f19f724]<https://baxter.com>
Hi
I'm trying to get dlg.list on a busy system via jsonrpc and xhttp.
I already increased:
tcp_conn_wq_max=2048
tcp_wq_max=20480
modparam("ctl", "binrpc_max_body_size", 2048)
modparam("ctl", "binrpc_struct_max_body_size", 2048)
But it looks like the tcp write queue overflows up at shy over 1MB
reply no matter what I change.
Any ideas what else to toggle?
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Hello,
The corex module manages extended flags with functions the setxflag(), isxflagset(), and resetxflag().
tmx.t_flush_xflags() does “flush the extended flags from the current SIP message into the already created transaction. It makes sense only in routing
block if the transaction was created via t_newtran() and the extended flags have been altered since.
It is not needed to execute this function when using t_relay() (or similar tm relay functions, xflags are synchronized automatically in that case).”
The documentation of the tm module does not say anything about extended flags - in particular it does not say that t_relay() synchronizes them.
Moreover, while the corex module utilizes the extended flags, the documentation of tm and tmx do not state, that they depend on the corex module.
So, what are extended flags?
Greetings
Дилян
Hello all! I have a question. In my system we use a series of VoIP
providers to receive external calls to our telephone numbers, for the same
provider we have several registered numbers and we need our providers to
send that information in some way, so far it does not seem that there is a
common way to do it (report the number that was dialed by the person who
made the call). From what I understand, SIP does not have a place for that
information because there are no things like phone numbers in the
specification, my question is, is there a correct place to send such
information, or is it entirely up to each provider? (Now, the most common
is to have in the RURI and To header, the id of the trunk registered in the
uacreg table) I have seen a couple of options that seem to me the most
“correct”: send the information in the To header or send in a custom header.
I really appreciate any help you can provide.
Atenciosamente,
[image: photo] <https://www.techer.com.br/>
*Carlos Escalona*
IT - Development
41 3073-0091 | R 1011
www.techer.com.br
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e, por favor, notifique o remetente.
At a quick glance it looks like using socket_workers when the listen parameter uses the interface name does not work. For example:
children = 4
socket_workers=2
listen=udp:10.10.10.10:5060
Gives two listeners on udp:10.10.10.10:5060
children = 4
socket_workers=2
listen=udp:eth0:5060
Gives four listeners on the socket.
Note this is just a quick cursory observation. Is it by design or a design limitation?
Kaufman
Hello,
Fosdem 2023 returns to Brussels, Belgium, for a physical event and I'm
writing to see if there are many people from the community going to the
event and want to organize again the usual RTC/Kamailio dinner on
Saturday evening (Feb 4, 2023).
Each person should pay for himself, likely to be some price increase
comparing with the last one.
If there is enough interest, Torrey will help to book a place to
accommodate us (first choice could be the same restaurant like in the
past, if available).
Reply if you want to join the dinner and say how many other people are
joining you.
Note that RTC Devroom got at this edition a half a day room allocation,
on the afternoon of Feb 5, 2023 (Sunday).
Cheers,
Daniel
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Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com
Hello,
please keep the list in CC.
Ok, the machine seems to not really using all its cores. Regarding playing a file, this can be also done with rtpengine:
https://kamailio.org/docs/modules/5.5.x/modules/rtpengine.html#rtpengine.f.…
Cheers,
Henning
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Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>
From: Mohammad Reza Keshavarzianpoor <keshavarzianpoorm(a)gmail.com>
Sent: Monday, January 16, 2023 3:35 PM
To: Henning Westerholt <hw(a)gilawa.com>
Subject: Re: [SR-Users] rtp lost package in 300 concurrent calls and above
thanks for your reply
my kamailio server has 8GB of ram and 4core*2.6 on esxi(dedicated)Ubuntu 22.04.1 LTS. is it low? htop on server shows load average 1.04 at max and I didn't see more than 10% load on any core of the server.
I've selected rtp media server because of playback of a voice file over incoming calls in the future usage
On Mon, Jan 16, 2023 at 5:45 PM Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> wrote:
Hello,
do you observe (too) high load on the system with the standard system management tools?
If you want to just proxy calls, my recommendation would be to use rtpengine. Here you should be able to achieve higher concurrent calls rates, if you have a few CPUs on a decent machine.
Cheers,
Henning
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Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>
From: Mohammad Reza Keshavarzianpoor <keshavarzianpoorm(a)gmail.com<mailto:keshavarzianpoorm@gmail.com>>
Sent: Monday, January 16, 2023 2:50 PM
To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>
Subject: [SR-Users] rtp lost package in 300 concurrent calls and above
Hi,
I've installed last version of kamailio over ubuntu. I need to broadcast machine calls to destinations. my concurrent call should be around 300(my pstn channels), so I start calls with cps around 20, because my calls contains a less than 30 second and if destination answer the call it will hanged up in less than or equal to 30 seconds.
My problem is: to test my kamailio config, I'm using startrinity sip tester over an win server 2019 on local network. and it shows about 20% lost package on rtp and a jitter about 500ms. what should I do? my config is based on rtp media server module. Is there a complete working sample config for kamailio with rtp media server module?
I tried rtpproxy and rtp engine in past and the same problem happened.
Thanks for any help you can provide
Best regards
Hi,
I've installed last version of kamailio over ubuntu. I need to broadcast
machine calls to destinations. my concurrent call should be around 300(my
pstn channels), so I start calls with cps around 20, because my calls
contains a less than 30 second and if destination answer the call it will
hanged up in less than or equal to 30 seconds.
My problem is: to test my kamailio config, I'm using startrinity sip tester
over an win server 2019 on local network. and it shows about 20% lost
package on rtp and a jitter about 500ms. what should I do? my config is
based on rtp media server module. Is there a complete working sample config
for kamailio with rtp media server module?
I tried rtpproxy and rtp engine in past and the same problem happened.
Thanks for any help you can provide
Best regards