Hi All,
I'm compiling the load_balancer module with some pretty minor changes
however the resulting load_balancer.so is ~500k however the standard
release is 120k - a size increase of 4x.
My question is simple - why is my version so much bigger? Are there
"make" flags that are used for the
If the backend servers are both the same instance then this seems to be the
correct behaviour?
I believe the probing is supposed to be a simple SIP response healthcheck
which applies to the destination globally (i.e. 1.2.3.4 is offline), the
groups are just a way of splitting up resources
issue you report here
> is also (in a more detailed way) reported in 3297 HG ticket, right ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
> https://www.siphub.com
>
> On 02.02.2024 16:53, Callum Guy
Hi All,
I'm implementing the load_balancer module on a very busy system where
thousands of calls may arrive in a matter of seconds. The module is
configured to receive heartbeats every 1 second from many freeswitch
servers, I have this set as a low value to try and keep OpenSIPs up to
date with
Hi Faheem,
This is a simple case of regex support - I believe OpenSIPs uses Extended
Regular Expressions and not Perl Compatible Regular Expressions.
Swap out \d with the longer [0-9] format and your expression should match.
Happy new year,
Callum
On Tue, 2 Jan 2024 at 11:27, Faheem Muhammad
I wanted to follow up with confirmation that opensips is behaving
normally here and an issue with delayed HTTP is in fact occurring
outside of the server.
My apologies for any confusion.
Best regards,
Callum
On Tue, 31 Oct 2023 at 16:21, Callum Guy wrote:
>
> Hi All,
>
> I'm se
Hi All,
I'm seeing a small number of sessions reporting timeouts for async
rest_client post requests. These occur at peak times for system load
and present the following error:
ERROR:rest_client:_resume_async_http_req: connected, but transfer timed out (5s)
The return code is -3 and HTTP
in the internal definition
> of the function.
>
> Would you be able to run a test if I'll get you a patch ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> https://www.siphub.com
>
> On 10/10/23 12:50 PM,
Hi All,
I can't find a function to reset a branch route. Instead I'm just creating
a new empty branch route which I set in the failure route but this seems
superfluous.
Am I missing something or does it simply not exist?
Callum
--
Voting link
The idea is that you create your own based on your unique
infrastructure requirements.
Learn you must.
https://www.opensips.org/Documentation/Manual-3-3
On Thu, 14 Sept 2023 at 15:21, Prathibha B wrote:
>
> I need the conf file for opensips 3.3
>
> Sent from Outlook for Android
>
Hi All,
I've observed a behaviour (OpenSIPs 3.2.4) where the contact stored
against an active dialog is not populated until the call is answered.
Using opensips-cli dlg_list I see the following:
Ringing - https://gist.github.com/spacetourist/2502f6b76a95bb2f500fda5291e1c93b
Answered -
> application server.
> >
> > However when a 183 comes back from the application server, the IP is
> updated to the internal address, regardless of which i/e combination I use.
> I can't seem to get it to populate the external IP.
> >
> > Thoughts?
> >
>
Thoughts?
>
> Gavin
>
> On Wednesday, April 19, 2023 at 12:52:04 PM ADT, Callum Guy
> wrote:
>
>
> Hi Gavin,
>
> Its been a while since i used rtpproxy (favouring RTPEngine these
> days) however I believe the main issue may be the listen option - this
> should
e same IP (from the application server)
> that comes in is the one that is sent back to the client. It should be the
> public IP address (142.1.2.3).
>
> Thoughts?
>
> Thanks,
>
> Gavin
>
>
> On Wednesday, April 19, 2023 at 07:06:54 AM ADT, Callum Guy
> wrote:
>
>
Hi Gavin,
Using an RTP proxy is a good approach, you'll need to set it up in
bridge mode so that it is aware of the internal and external addresses
so that it can present the public IP to the client and private to the
application server.
You probably don't need fix_nated_sdp as
Hi All,
I'm looking for a clean way to de-register a WS location when I
receive a REGISTER with expires set as a contact param:
;expires=0
The issue is that a ul record is stored with the following contact and
the parameters are included in the matching process for registrar
module remove()
Hi James,
It could certainly be clearer!
Here's an extract from my script and some example inserts for a client and
server record.
https://gist.github.com/spacetourist/788ea722901e81d355850842e2b17cda
INSERT INTO opensips_dev.tls_mgm (id, domain, match_ip_address,
match_sip_domain, type,
Hi Daniel,
I believe you're looking for this feature as included since 3.3
https://www.opensips.org/Documentation/Interface-StatusReport-3-3
Enjoy,
Callum
On Fri, 24 Mar 2023 at 14:19, Daniel Zanutti
wrote:
> Hi
>
> Is there a way to check the status of initial loading of routes, on the
>
Hi Matt,
You are not alone, I have just performed the same update and ran into the
same problem!
No config changes, just an opensips package update on a CentOS 7 server.
My only lead so far is the server referring to a stale nonce in its reply,
this may be a red herring as I reverted so quickly
! The affected
system is running 3.2.4 - happy to upgrade if there has been any work in
this area!
Thanks,
Callum
On Wed, 22 Feb 2023 at 21:40, Callum Guy wrote:
> Hi All,
>
> I operate OpenSIPs in front of a bank of FreeSWITCH instances and I'm
> currently seeing an issue with FreeSWITCH cra
Hi All,
I operate OpenSIPs in front of a bank of FreeSWITCH instances and I'm
currently seeing an issue with FreeSWITCH crashing during a SRTP DTLS
renegotiation triggered by a RE-INVITE.
I've tracked this down to a WSS registrar which is issuing
rtpengine_offer(...) for the INVITE and any
Hi All,
*(originally sent back in December with an image embedded which possibly
got the mail rejected!)*
I'm experiencing a memory leak on an OpenSIPs 3.2.10 instance (was 3.2.4,
upgrades to 3.2.9 and 3.2.10 did not resolve) which is acting as a WebRTC
proxy - this is running on AlmaLinux 9.0
Added the worlds simplest PR for you!
https://github.com/OpenSIPS/opensips/pull/2958
On Mon, 28 Nov 2022 at 11:17, Callum Guy wrote:
> Looks like a typo in the module!
>
>
> https://github.com/OpenSIPS/opensips/blob/master/modules/rate_cacher/rate_cacher.c#L127
>
>
Looks like a typo in the module!
https://github.com/OpenSIPS/opensips/blob/master/modules/rate_cacher/rate_cacher.c#L127
I suggest trying "cients_hash_size" until patched :)
On Mon, 28 Nov 2022 at 08:19, Wadii ELMAJDI | Evenmedia
wrote:
> Hello,
>
> I was able to install ratecacher module and
istration time you could save the User Agent in the attributes per
> registration [1] and at INVITE time you can check that in branch_route and
> make your decision there.
>
> [1]
> https://opensips.org/html/docs/modules/3.2.x/registrar.html#param_attr_avp
>
>
> Regards,
&
Hi All,
I'm working on a project that requires me to evaluate the user agent of a
registered contact before making a decision on a current registration
attempt.
Is there a method to do this natively? If not, what is the best approach?
I haven't found anything yet, the current options i'm
Hi Team,
I just wanted to report an incorrect behaviour for a comment from the dev
team.
I'm operating an RTC registrar on 3.2.4 and defining that each device (web
console) can have a maximum of one active contact, overridden by any new
registration for that AoR. The previously active devices
Hi All,
I have configured my registrar with max_contacts 1, allowing
subsequent registrations from that contact to overwrite.
I am looking to intercept the de-registration and send a message to the
losing contact. Ideally I would use the existing SIP connection to send it
before termination, the
Excellent, another string to the bow!
On Thu, 24 Feb 2022 at 10:28, HS wrote:
> Callum.
>
> Thanks a lot for the help. I seem to have been able to compile the module
> and add it. Now working on port 443 on apache2 + Opensips. The fun never
> ends :D
>
Manually compiling modules is relatively straightforward if you've got the
build tools installed. I can't provide a copy as I'm using CentOS.
The process is simply to clone the repo on your target system (or
equivalent), checkout the release tag and then compile as shown below
(after adjusting to
Depending on your OS version, looks like 3.0 repo has been removed:
https://apt.opensips.org/dists/buster/
If so you'll either need to find a copy or compile it yourself (or upgrade!)
On Wed, 23 Feb 2022 at 09:49, HS wrote:
> Apologies, I have already tried:
>
> apt-get install
You'll probably want to ensure you have the following packages installed,
presuming you are using the provided repo's:
opensips-wss-module
opensips-tls-module
opensips-tlsmgm-module
On Wed, 23 Feb 2022 at 09:08, HS wrote:
> Dear all.
>
> Trying to following the tutorials, however, when I
Hi All,
I'm wondering if anyone in the community has encountered the iOS 15 WebRTC
bug as reported here:
https://www.wowza.com/community/t/ios-15-webrtc-problem/93868/4
https://developer.apple.com/forums/thread/689293
The bug causes INVITES to fail to be digested by opensips with a log
message
You could try to install the required version?
https://centos.pkgs.org/7/okey-x86_64/automake-1.14-1.el7.x86_64.rpm.html
On Mon, 29 Nov 2021 at 11:52, Virendra Bhati via Users <
users@lists.opensips.org> wrote:
> Dear Team,
> I am not able to install Opensips 3.2. I am facing an issue with
Try this:
$var(result) = "$avp(id_asterisk)|$avp(codecs)";
On Mon, 18 Oct 2021 at 10:10, Alain Bieuzent wrote:
> Hi All,
>
>
>
> I’m’ trying to concatenate two avp result to one string
>
> The first avp(id_asterisk) is an integer,
>
> the second $avp(codecs) is a string.
>
>
>
> When I try to
ram("tls_mgm", "tls_method_col", "method")
> modparam("tls_mgm", "verify_cert_col", "0")
> modparam("tls_mgm", "require_cert_col", "0")
> modparam("tls_mgm", "certificate_col", "certif
You need to insert the certificate contents into the table rather than a
file, to my understanding - you'd probably want to convert the certs to PEM
format prior to doing this.
On Tue, 25 May 2021 at 14:20, Saurabh Chopra wrote:
> Hi Opensips Team/Razvan,
>
> I am using the TLS_MGM module
subst_uri only works on the request uri, try again with subst()!
On Mon, 17 May 2021 at 08:58, Miha via Users
wrote:
> Hello
>
> i need a little help how to chnage RR in responses to UDP GW (requestes
> goes via TLS to MS teams).
>
> So in reply i have like this:
> RECORD-ROUTE:
> ,
> .
>
>
Thinking outside the box here but... maybe update to 3.1.1?
https://www.opensips.org/Documentation/Migration-3-0-0-to-3-1-0
On Wed, 14 Apr 2021 at 17:35, HS wrote:
> Bogdan-Andrei,
>
> Thanks a lot for sharing these links. I had actually looked at them
> previously, all are for Opensips 3.1 -
Excellent, thanks for beginning this to my attention - will track.
On Wed, 14 Apr 2021 at 11:52, Alexey Vasilyev
wrote:
> Hi, yes there is an issue.
>
> Please follow here: https://github.com/OpenSIPS/opensips/issues/2433
>
>
>
>
> -
> ---
> Alexey Vasilyev
> --
> Sent from:
>
Hi All,
I recently encountered an issue where our certificates were renewed,
following which I issued: *opensips-cli -x mi tls_reload*
The CLI action indicated success however on closer inspection of the
handshake we could see the previous certificate was continuing to be
presented. Previously I
You should try setting the min expires value, the documentation states that
any values lower than this will be overridden with the minimum you define!
https://opensips.org/html/docs/modules/3.0.x/registrar.html#param_min_expires
On Wed, 17 Mar 2021 at 06:43, Jeffrey Zhao wrote:
> Dear Team
>
>
My advice would be to get stuck in, I had plenty of questions after reading
the release blog posts but once I started implementing it all made sense.
Anything specific concerning/confusing you at this stage?
Good luck!
On Wed, 17 Mar 2021 at 09:46, Mark Allen wrote:
> Thanks Johan - I can see
Hi Joseph,
I haven't fully digested your scenario however you may have some luck using
the nathelper function fix_nated_contact - presuming NAT is not an issue.
https://opensips.org/html/docs/modules/3.1.x/nathelper.html#func_fix_nated_contact
Otherwise you'll probably be able to achieve this
S IP address but rather to an address on the local LAN -
> hence the problem. Thanks for your help.
>
> Cheers,
>
> Mark
>
>
>
> On Wed, 10 Mar 2021 at 09:26, Callum Guy wrote:
>
>> Hi Mark,
>>
>> It sounds like you may be having issues with the pr
Hi Mark,
It sounds like you may be having issues with the proxy not keeping itself
in path for certain call scenarios.
Are you able to provide a SIP trace and/or opensips config? Also if you're
running Blink on a Linux system, can you get a SIP trace there to see if
the BYE is being generated
n, 15 Feb 2021 at 15:42, Callum Guy wrote:
>
> Hi All,
>
> Running 3.1 release.
>
> I'm trying to implement a proxy which *only* supports PN-enabled
> devices however I'm running into some implementation issues for my use
> case.
>
> My project is targeting a v
Hi All,
Running 3.1 release.
I'm trying to implement a proxy which *only* supports PN-enabled
devices however I'm running into some implementation issues for my use
case.
My project is targeting a very low usage subscriber base (users may be
idle for months) so I'd like to disable the timer
Thanks for that Vlad, I always learn something from these article
releases and this was no exception
On Thu, 11 Feb 2021 at 12:35, Vlad Patrascu wrote:
>
> Hello everyone,
>
> Check out this new blog post [1] that recounts our process of searching
> for and choosing a SSL/TLS library to use in
Maybe permissions or similar - have you tested the example from the docs?
exec("ls -l", , "$var(out)", "$var(err)", "$avp(env)");
xlog("The output is $var(out)\n");
xlog("Received the following error\n$var(err)");
On Thu, 4 Feb 2021 at 10:36, Dragomir Haralambiev wrote:
>
> Hello,
>
> I try to
An alternative option would be to leverage cachedb_local and opensips-cli
to implement your list of accounts and rate limits. It has the advantage of
using the internal opensips cache service and is probably your most high
performance option, with the CLI you can automate data refreshes using
I believe the DB followed by a dr_reload is your only option...
On Fri, 30 Oct 2020 at 15:49, Mark Farmer wrote:
> Hi everyone
>
> I am looking for a way to manage routes etc in drouting without using
> opensips-cp.
> I was hoping to find a MI function to add/remove routes but there only
>
Have you double checked it's not a firewall issue on the SIP proxy?
The transport is typically UDP so there is a fair chance it's blocked.
On Tue, 27 Oct 2020 at 19:05, John Quick wrote:
> Using OpenSIPS v2.4.8 and the latest release of rtpengine. The same was
> also
> happening with some
wrote:
>
> On 20.08.2020 17:46, Callum Guy wrote:
> > I presume this is fallout from a recent network issue so I plan to
> > restart all instances during a quiet period which I'm sure will
> > resolve it.
> >
> > Is there any change that visibility of the async stac
Hi All,
Noticed these in my logs suggesting that something has jammed in the
async tables:
2020-08-20T15:42:24.145805+01:00 FR-P-SIPSBC-1 opensips[240129]:
WARNING:rest_client:get_multi: max async transfers! (250)
2020-08-20T15:42:24.146290+01:00 FR-P-SIPSBC-1 opensips[240129]:
it chooses which one to use for
conditional statements etc
On Thu, 20 Aug 2020 at 07:35, Igor Pavlov wrote:
>
> Thanks a lot for {s.int} ! I forgot that my dialog value is string.
> Transforming to int helped.
>
>
> ср, 19 авг. 2020 г. в 02:33, Callum Guy :
>>
>> Have you mat
Have you matched the dialog before running this check? Just wondering
if one of those values is stale, do the durations match up with
reality for the example calls?
Also maybe rule out type issues with $(dlg_val(dialog_min_time){s.int})
On Tue, 18 Aug 2020 at 17:35, Igor Pavlov wrote:
>
> Hi
Thanks Johan, exactly what the doctor ordered!
Most appreciated
On Mon, 17 Aug 2020 at 14:26, Johan De Clercq wrote:
>
> use t_relay wih 0x2 option.
>
> On Mon, Aug 17, 2020, 15:16 Callum Guy wrote:
>>
>> Hi All,
>>
>> Using OpenSIPs 3.0.3
>>
>>
Hi All,
Using OpenSIPs 3.0.3
I'm dealing with a client device with a faulty network, they are using a
softphone WebRTC client and the TCP connections disappear sporadically.
When the media server issues a RE-INVITE session timer OpenSIPs discovers
the closed TCP connection and returns 477 to
Yes this is very much achievable and a common topology.
You'll need to look into media proxy software (RTPEngine/rtpproxy/etc)
as well but this can run on the same device if sufficient resources
are available.
OpenSIPs is a great choice!
On Wed, 12 Aug 2020 at 15:00, Adam Obuchowski wrote:
>
That'd be great, hope this comes together!
On Tue, 11 Aug 2020 at 14:40, Maxim Sobolev wrote:
> Interesting work Adrian! Any chance you can be interested in coming over
> to our SIP Chronicles videocast to talk about it and perhaps do a live
> demo?
>
>
It is not questions I seek, but answers to the great mystery.
On Tue, 21 Jul 2020 at 07:57, Alexey Kazantsev via Users <
users@lists.opensips.org> wrote:
> Hello,
> So what is the question?
> ___
> Users mailing list
> Users@lists.opensips.org
>
Hi Rob,
I'm interested to follow your thread to hear more about this, I have
found that some flags are valid yet undocumented during initial setup
of some RTC compatable proxies.
Two in particular: DTLS-passive and SDES-disable both of which appear
to influence behaviour of RTPEngine in
OpenSIPs 3.1 and Nasa DM-2 (https://www.nasa.gov/specials/dm2/) on the same
day? Awesome!
On Thu, 14 May 2020 at 10:02, Bogdan-Andrei Iancu
wrote:
> Hi all,
>
> We planned an ambitious roadmap [1] for OpenSIPS 3.1, but we were even
> more ambitious by trying to complete it. It was a long way,
Do you have the module loaded? Have you
installed opensips-python-module.x86_64? Confirm in your module path that
it exists!
loadmodule "python.so"
On Wed, 13 May 2020 at 03:18, Gordon Yeong wrote:
> Anyone know what's going on?
>
> Gordon
>
>
> On Tue, 12 May 2020 at 17:10, Gordon Yeong
Sippy cup has some good media generation capabilities, still using sipp
under the hood.
https://mojolingo.github.io/sippy_cup/
On Mon, 11 May 2020 at 18:45, Tomi Hakkarainen wrote:
> I agree
>
> BR, Tomi
>
> On 11. May 2020, at 19.48, johan wrote:
>
> hmmm sipp with your own rtp files.
> On
Hi All,
I've had a couple of incidents this week where OpenSIPs crashed during BYE
processing, in both cases the SIP logs show BYE arriving from both sides of
a call within a couple of milliseconds - these overlap and result in 481's
being both sent to our media server and being received from our
Hi All,
Some of our clients are brave enough to access our OpenSIPs WebRTC
gateway using Microsoft Edge.
We've had some teething issues which have been diagnosed as a failed
SNI check due to the character casing, our certificate presents common
and alt names in lowercase (i.e. rtc.opensips.org)
uot;pkmem:19-real_used_size": 12794072,
"pkmem:20-real_used_size": 12801088,
"pkmem:21-real_used_size": 12794264,
"pkmem:22-real_used_size": 12796232,
"pkmem:23-real_used_size": 12797184,
Thanks for reading, I am curious to unders
gt;
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
>
> On 4/28/20 1:38 AM, Callum Guy wrote:
> > Hi Bogdan,
> >
> > I'm still searching for my memory leak, just downgraded from 3.0.2 to
> > 3.0.1 and should be able to confirm if that has
Hi All,
I've been hunting a minor memory leak in my config and wanted to check in
with the devs in case it is related to ues of the parameter:
disable_503_translation=yes
Here is the implementation link to save you a few seconds:
Hi Sasmita,
I would advise that you capture this information in a branch flag during
registration, these will be stored in location and retrieved when
performing a matching lookup().
So:
if ($pr == "ws" || $pr == "wss") {
setbflag(SRC_WS);
}
save("location_table")
When you do the lookup
That'll do it..
On Fri, 17 Apr 2020 at 16:00, Mark Farmer wrote:
> OK, fixed it.
>
> Turned out to be this breaking it by overwriting $acc_extra(customer_id)
> with a blank value.
>
> ...
> else $acc_extra(Call_Flow) = "Internal";
> $acc_extra(customer_id) = $var(rule_attrs);
>
You're not the only one my friend, I've seen plenty of discussions on the
topic in the mailing list so browse the archives for details.
i.e.
https://opensips.org/pipermail/users/2018-September/039895.html
Here's a quote from Bogdan later in that thread:
"If there is no load (worker processes
isflagset(RTPENGINE_ENGAGED)) {
rtpengine_delete();
}
}
On Mon, 16 Mar 2020 at 13:33, Callum Guy wrote:
>
> Hi Ben,
>
> Thank you for the information, I've checked the tm module docs and
> t_check_trans() doesn't highlight this behaviour - it just sounds like a
> me
me.
>
>
>
> [1] - https://github.com/OpenSIPS/opensips/issues/1637
>
>
>
> Ben Newlin
>
>
>
> *From: *Users on behalf of Callum Guy <
> callum@x-on.co.uk>
> *Reply-To: *OpenSIPS users mailling list
> *Date: *Monday, March 16, 2020 at 7:23 A
Hi All,
I have a simple question regarding availability of AVP variables in CANCEL.
I'm not sure when OpenSIPs will load the AVP's for a transaction so am
looking for information here. The situation is that I want to flag sessions
using a media proxy and close the sessions when a CANCEL arrives
I encountered a similar issue recently, I was using dialog variables
to flag sessions where RTPEngine is engaged so rtpengine_delete only
fired on applicable BYE/CANCEL requests. For reasons I have not yet
understood the dialog variable was not always available so the
sessions were left open and
Hi Jehan,
Sounds like you want to be using fix_nated_contact() - when the INVITE
arrives you can try the following:
# Check if contact is RFC1918
if (nat_uac_test(1)) {
# Replace the contact IP with the received address from the network
fix_nated_contact();
}
If you look at the
That is the same as the example above!
On Fri, 28 Feb 2020 at 10:11, johan wrote:
> no, what I mean is this: if lookup_location(...) needs to be if
> (lookup_location())
> On 28.02.20 11:06, Grant Bagdasarian wrote:
>
> Hi Johan,
>
> I’ve been testing with a nightly build (not the latest) these
is not included in any
tagged releases at this time so I will amend the `ping_timeout` as
suggested.
Thanks for the invaluable insight.
On Wed, 19 Feb 2020 at 14:29, Liviu Chircu wrote:
> Hi, Callum!
>
> On 18.02.2020 15:56, Callum Guy wrote:
>
>
> INFO:usrloc:receive_ucontact_inser
Hi All,
I'm running a full sharing cluster for hot standby purposes and have been
noticing that the backup node will periodically report the messages below.
INFO:usrloc:receive_ucontact_insert: failed to fetch local urecord -
creating new one (ci: '0_751733367@10.0.0.13')
Hi Grant,
There could very well be a better way so hopefully someone else will chime
in if they have a better solution however I would leverage a simple
$var(example_mode)
= 'reply'; - $var's operate on a per process basis so you would simply set
it before the route(GEN) call and check its value
Hi,
All replies arrive at a separate route block, configurable during
processing of the parent request.
You can define many reply processing routes and allocate requests to these
at your discretion using t_on_reply("example_a"); or
t_on_reply("example_b"); etc
Docs are here:
I would imagine Liviu is correct, the answer will be quite clear
Not sure if you have seen my earlier comments but wanted to reiterate that
your initial email shows a resource limit error "Starting opensips (via
systemctl): Job for opensips.service failed because a configured resource
limit was
:15 Callum Guy, wrote:
> You're on Centos 7, systems my friend.
>
> Try systemctl start opensips
>
> Otherwise put the service file in systemd tree, an example exists on
> GitHub, and run systemctl daemon-reload before trying to start again.
>
> If it still doesn't work you'
You're on Centos 7, systems my friend.
Try systemctl start opensips
Otherwise put the service file in systemd tree, an example exists on
GitHub, and run systemctl daemon-reload before trying to start again.
If it still doesn't work you'd need to check that you have the opensips
binary in the
nSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 1/24/20 3:27 PM, Callum Guy wrote:
> > Hi Razvan,
> >
> > Just wondering if the below commit can/will be included in the 3.0.2
> > release? I don't see a 3.0.2 branch or tag on Github so I'm not s
Hi Razvan,
Just wondering if the below commit can/will be included in the 3.0.2
release? I don't see a 3.0.2 branch or tag on Github so I'm not sure how to
check!
https://github.com/OpenSIPS/opensips/commit/7b9239d63f412a1194e10c97611489d5facfdf74
Thanks,
Callum
On Thu, 23 Jan 2020 at 09:58,
, I couldn't figure out from where I can get this kind of
> .crt files.
>
> On Mon, Jan 20, 2020 at 11:49 AM Callum Guy wrote:
>
>> Hi Ali,
>>
>> You'll need to setup your cipher list and DH file. You can generate a DH
>> param file like this: *openssl dhparam -o
ps-solutions.com
>
> OpenSIPS Summit, Amsterdam, May 2020
>www.opensips.org/events/Summit-2020Amsterdam
> OpenSIPS Bootcamp, Miami, March 2020
>www.opensips.org/training
>
> On 09.12.2019 13:13, Callum Guy wrote:
> > Hi All,
> >
> > I wanted to follow up on
Hi Ali,
You'll need to setup your cipher list and DH file. You can generate a DH
param file like this: *openssl dhparam -out dhparam.pem 4096*
If you want to review locally available cipher suites you can run: *openssl
ciphers -v*
The OpenSIPs documentation clarifies the module configuration
vate IPs.
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
> OpenSIPS Summit, Amsterdam, May 2020
>https://www.opensips.org/events/Summit-2020Amsterdam/
> OpenSIPS Bootcamp, Miami, March 2020
>h
Hi All,
I am operating a registrar which proxies calls to an internal network
of media servers.
Most of my subscribers are operating using RFC1918 addresses behind
NAT. We detect this configuration through nat_uac_test() and patch the
SIP using fix_nated_contact(). By rewriting the requests the
Hi All,
I wanted to follow up on a recent issue I experienced to understand if
it was due to user error or a bug that needs to be patched.
The issue was traced back to a simple function call in the permissions module:
check_source_address(0, $avp(address_desc))
Nearly every request processed
ght in your spam bin for being too large
On Tue, 3 Dec 2019 at 17:31, Johan De Clercq wrote:
>
> I think you can. Check the documentation of rtpengine on github. And if you
> can, please use the latest commit.
>
> On Tue, 3 Dec 2019, 18:02 Callum Guy, wrote:
>>
>> Hi A
Methods are almost interchangeable though - check out the docs:
https://opensips.org/html/docs/modules/3.0.x/rtpengine.html
On Wed, 4 Dec 2019 at 21:29, David Villasmil
wrote:
> are you setting up rtpENGINE or rtpPROXY?
> They're not the same...
>
> Regards,
>
> David Villasmil
> email:
.
Callum
On Tue, 3 Dec 2019 at 17:31, Johan De Clercq wrote:
> I think you can. Check the documentation of rtpengine on github. And if
> you can, please use the latest commit.
>
> On Tue, 3 Dec 2019, 18:02 Callum Guy, wrote:
>
>> Hi All,
>>
>> I'm working throug
Why not just glue together the strings?
$var(a) = "972" + $rU;
Seems pretty effective to me :)
On Tue, 3 Dec 2019 at 22:02, VOIP Security via Users <
users@lists.opensips.org> wrote:
> Hi,
>
> I am struggling with openSIPS regex rules to append some prefix before
> regex. I have this regex
5:flagsl11:initialized4:send4:recv15:ICE
controllinge6:totalsd3:RTPd7:packetsi4063e5:bytesi698836e6:errorsi0ee4:RTCPd7:packetsi17e5:bytesi2248e6:errorsi0eee6:result2:oke
On Sat, 30 Nov 2019 at 22:51, Callum Guy wrote:
> Hi Ben,
>
> Thank you for your reply and insight here, very helpful to kn
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