Juha Heinanen writes:
> I just tried by replacing ca_list file of my proxy (that contained ca
> certs of my peers) with a single bogus ca cert. Then I executed tls.cfg
> and made a call from one of the peers to my proxy. My proxy still
> recognized the call as coming from the peer based on its t
On 01/08/2016 04:32 PM, Benjamin Fitzgerald wrote:
I think #1 fixed it for me! Thank you so much! I changed the RTP timeout
on a test account SIP account and immediately it resolved the issue.
Excellent! Happy to help.
You're right, sending a BYE would effectively synchronize them
however I
Hi Alex,
Thank you! Your suggestion will most likely fit our solution.
/V
On Fri, Jan 8, 2016 at 4:42 PM, Alex Balashov
wrote:
> On 01/08/2016 04:40 PM, Vik Killa wrote:
>
> That last statement was in-accurate. Im not trying to modify the R-URI
>> at all actually.
>> I'd like to create a variabl
On 01/08/2016 04:40 PM, Vik Killa wrote:
That last statement was in-accurate. Im not trying to modify the R-URI
at all actually.
I'd like to create a variable.
Right, you're trying to extract a value from the RURI, transform it, and
copy the transformed value into something else.
I think my
On 01/08/2016 04:38 PM, Vik Killa wrote:
I'm not trying to replace the R-URI like in your example, im trying to
remove a prefix from the RURI
Oh, I see.
You might consider stripping[1] the necessary number of characters from
the user part of the RURI, then.
$var(prefix_len) = $(var(Pref
That last statement was in-accurate. Im not trying to modify the R-URI at
all actually.
I'd like to create a variable.
On Fri, Jan 8, 2016 at 4:38 PM, Vik Killa wrote:
> I'm not trying to replace the R-URI like in your example, im trying to
> remove a prefix from the RURI
>
> On Fri, Jan 8, 2016
I'm not trying to replace the R-URI like in your example, im trying to
remove a prefix from the RURI
On Fri, Jan 8, 2016 at 4:34 PM, Alex Balashov
wrote:
> On 01/08/2016 04:28 PM, Vik Killa wrote:
>
> $var(destnumber1) =
>> $(ru{re.subst,/^sip:$var(PrefixMatch)(.*)@(.*)/\1/});
>>
>
Assuming the answer is no this is not possible, then does anyone know of a
way to accomplish this? Perhaps with textops module?
Thanks,
/V
On Fri, Jan 8, 2016 at 4:30 PM, Alex Balashov
wrote:
> On 01/08/2016 04:28 PM, Vik Killa wrote:
>
> $var(destnumber1) =
>> $(ru{re.subst,/^sip:$
On 01/08/2016 04:28 PM, Vik Killa wrote:
$var(destnumber1) =
$(ru{re.subst,/^sip:$var(PrefixMatch)(.*)@(.*)/\1/});
But, it's certainly worth asking if what you're trying to accomplish
here can't be accomplished differently...
$var(destnumber1) = "$rz:" + $var(PrefixMatch) +
Alex,
I think #1 fixed it for me! Thank you so much! I changed the RTP timeout on
a test account SIP account and immediately it resolved the issue.
You're right, sending a BYE would effectively synchronize them however I
did not think keepalive using OPTIONS scheme would send a BYE message in
the
On 01/08/2016 04:28 PM, Vik Killa wrote:
$var(destnumber1) =
$(ru{re.subst,/^sip:$var(PrefixMatch)(.*)@(.*)/\1/});
$var(destnumber2) = $(ru{re.subst,/^sip:00(.*)@(.*)/\1/});
xlog("L_INFO", "destnumber1 $var(destnumber1)\n");
xlog("L_INFO", "des
Hello,
Is it possible to use $avp() or $var() inside re.subst?
Example:
$var(PrefixMatch) = "00";
$var(destnumber1) =
$(ru{re.subst,/^sip:$var(PrefixMatch)(.*)@(.*)/\1/});
$var(destnumber2) = $(ru{re.subst,/^sip:00(.*)@(.*)/\1/});
xlog("L_INFO", "des
Benjamin,
On 01/08/2016 03:25 PM, Benjamin Fitzgerald wrote:
1. Sorry to be unclear, the Asterisk channel does not stay up
indefinitely. We do have a max timeout but since a large portion of our
business is based on conference calling, the timeout is rather large. I
will definitely change the R
Hi Alex,
Thanks for your quick response.
1. Sorry to be unclear, the Asterisk channel does not stay up indefinitely.
We do have a max timeout but since a large portion of our business is based
on conference calling, the timeout is rather large. I will definitely
change the RTP timeout as my first
Hi Benjamin,
To some extent, this is just a perennial, existential problem of using a
proxy, so part of the answer is going to be that you need fundamentally
reliable signalling, speaking from the vantage point of something which
operates are a signalling relay (i.e. Kamailio).
However, I un
RTP timeout in asterisk is the best place to handle the situation. Another
option is SIP session timer, but it could give false negatives with NATed
clients.
On Friday 08 January 2016 11:56:51 Benjamin Fitzgerald wrote:
> Hi,
>
> I'm wondering what the best approach to handling a SIP dialog whe
Daniel-Constantin Mierla writes:
> Afaik, tls.cfg can be reloaded at runtime, that should reload the tls
> certificates linked there. Have you tried and it doesn't work?
>
> http://www.kamailio.org/docs/modules/stable/modules/tls.html#tls.r.tls.reload
I just tried by replacing ca_list file of my
Hi,
I'm wondering what the best approach to handling a SIP dialog when one
endpoint disappears/fails to send the BYE message.
I have Kamailio as a proxy for all mobile (iPhone/Android) SIP clients.
Occasionally, the user hangs up the call but no BYE message is received.
This means that Asterisk h
Welcome - glad to hear it was sorted out!
Cheers,
Daniel
On 08/01/16 18:32, Daniel W. Graham wrote:
>
> I follow now :) tested and working.
>
>
>
> Thanks Daniel for the help!
>
>
>
> -Dan
>
>
>
> *From:*Daniel-Constantin Mierla [mailto:mico...@gmail.com]
> *Sent:* Friday, January 8, 2016 3
Hello,
thanks -- one more thing, though: can you export the document as pdf to
be able to view it easy on different OSes as well as browsers?
Cheers,
Daniel
On 08/01/16 17:47, Franz Edler wrote:
>
> Hello Daniel,
>
>
>
> the short description is as follows:
>
>
>
> It is an “IMS in one box”
I follow now :) tested and working.
Thanks Daniel for the help!
-Dan
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: Friday, January 8, 2016 3:33 AM
To: Daniel W. Graham ; Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Audio issue when using 2 port ATA
You need to
Hello Daniel,
the short description is as follows:
It is an "IMS in one box" configuration, where I re-built the default
configuration of the original OpenIMSCore.
The configuration uses only the core-functions of the IMS. I omitted
(disabled) advanced functions like NAT, RTP-relay, antifl
Welcome - enhancements to the docs are always more than welcome!
Cheers,
Daniel
On 04/01/16 09:29, Phil Lavin wrote:
>
> Thanks, Daniel. I’ll update the docs to clarify this, if I get a moment.
>
>
>
>
>
> Cheers
>
>
>
> Phil
>
>
>
> *From:*sr-users [mailto:sr-users-boun...@lists.sip-rout
Hello,
as a intermediary summary, so far the upper limit is like 18 people
(announced as possible participants directly or indirectly on mailing
lists) and probably at least 10.
So maybe we should try to get a reservation in advance -- we will wait
more to see if others intend to join, but we nee
It is always possible to propose new feature, but not guaranteed they
will be in a release.
Usually, not to forget about them, we used issue tracker to collect
these proposals. However, if there are going to be many of them, we may
end up with a overloaded tracker with new feature requests. In tha
Oh, is it time to wish for things? :-)
What we would like to see is the ability to globally pause all OPTIONs
checks (set ds_ping_interval temporarily to 0 at runtime).
We have a master/slave setup and the slave is failing its pings because
it has no network access and it fills the log with error
On 08/01/16 09:59, Juha Heinanen wrote:
> Daniel-Constantin Mierla writes:
>
>> replying on this announcement to get it fresh in mind for everyone and,
>> if it is needed, start relevant discussions for upcoming major release
>> 4.4.
> I too would like to have capability to reload tls certificate
Load balancer (dispatcher) and webrtc (websocket) have example of
configuration in their documentation (module readme).
Integration with other application is approached on many web
articles/blogs. There are many ways of doing it, specific for various
use cases.
Cheers,
Daniel
On 08/01/16 10:09,
I would like to have capability to have basic functionality file in
kamailio.cfg and other integration and to have separated config file
for e.g. asterisk integration, load balancing, and webRTC and many more.
> Date: Fri, 8 Jan 2016 10:59:55 +0200
> To: mico...@gmail.com; sr-...@lists.sip-route
Daniel-Constantin Mierla writes:
> replying on this announcement to get it fresh in mind for everyone and,
> if it is needed, start relevant discussions for upcoming major release
> 4.4.
I too would like to have capability to reload tls certificates without
restarting Kamailio.
-- Juha
Hello,
besides Alex' relevant remarks on time interval for sampling, note that
getting high performances require tuning other parameters, typically
across many modules. For pipelimit, be sure you increase the value for
hash_size
http://www.kamailio.org/docs/modules/stable/modules/pipelimit.html#p
Hello,
replying on this announcement to get it fresh in mind for everyone and,
if it is needed, start relevant discussions for upcoming major release 4.4.
Cheers,
Daniel
On 14/12/15 09:07, Daniel-Constantin Mierla wrote:
> Hello,
>
> sketching the road to the next major release, so people can pl
Hello,
just to complete a bit about Via vs Record-Route: the reply received by
Kamailio will have only the addresses of the hops from Kamailio to the
sender of the request (caller). But if there are hops between Kamailio
and callee, those addresses are no longer in Via headers. Via is used to
rout
Hello,
if you need to change the routing logic, then you have to restart. If
you use UDP, it is not noticed at all by the clients and active calls
are not affected at all.
Restart is very fast. Also, the content of the config should be only
logic, the values (like IP addresses), should be inside
You need to engage branch route again in failure route. All those tm
route blocks need to be re-engaged for each t_relay().
Cheers,
Daniel
On 07/01/16 22:09, Daniel W. Graham wrote:
>
> The SDP was updated with RTPProxy IP.
>
>
>
> Yes, config was written around the default config, here are som
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