> background jobs running that send a lot of queries to the management
> interface.
>
> Thanks.
>
> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
&
ips -P
>> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>>
>> 14749 ?S 0:00 /usr/sbin/opensips -P
>> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>>
>> 14750 ?S 0:00 /usr/sbin/opensips -P
&
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
--
Callum Guy
Head of Information Security
X-on
--
*0333 332 | www.x-on.co.uk <http://www.x-on.co.uk> | **
<https://www.linkedin.com/company/x-on> <https://www.facebook
need to get rtpproxy working in bridging mode such that it opens two
sessions, one for each leg, each traversing NAT. This article may help:
https://saevolgo.blogspot.co.uk/2013/08/rtpproxy-revisited-kamailio-40.html
On Tue, Mar 6, 2018 at 2:29 PM Callum Guy wrote:
> Ah, zee NAT, sac
1236-10aa-d0431e9b59*
*Wireshark Failure:*
https://ibb.co/ici8Ny
*Wireshark Working:*
https://ibb.co/kePF2y
--
Callum Guy
Head of Information Security
X-on
--
*0333 332 | www.x-on.co.uk <http://www.x-on.co.uk> | **
<https://www.linkedin.com/company/x-on> <https://w
ion?
Thanks
On Mon, Jun 18, 2018 at 1:15 PM Callum Guy wrote:
> Hi All,
>
> I'm running a TLS protected service using OpenSIPs 2.3.3
> and openssl-1.0.2k-12.el7.x86_64 on a CentOS 7.5 server. RTPProxy is in
> place to forward all the media on every call.
>
> OpenSIPs
ng this, I suggest we increase the
> tls_mgm's module "tls_send_timeout" to some ridiculous value (e.g. 2000 ms)
> and see if the client write timeouts still happen.
>
> Best regards,
>
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 18.0
the necessary ports and that this is
something which can be addressed?
Finally if anyone has any other advice/articles regarding using OpenSIPs in
this way (dialog failover, using 2.4 features etc) then this would be
gratefully received!
Many thanks,
Callum
--
Callum Guy
Head of Infor
a
simpler approach out there (
https://github.com/ngvoice/rtpengine-redis-plugin)
Best Regards,
Callum
On Wed, Jun 27, 2018 at 11:45 AM Callum Guy wrote:
> Hi All,
>
> I am looking into updating my network to support live failover between two
> OpenSIPs instances acting as an
re.net
<http://pci-fram-proxy2.x-onsecure.net> opensips-m4cfg[30867]: Aug 8
08:55:10 [30867] ERROR:core:main: failed to solve module dependencies*
For now I have simply reverted back however I wonder if this is a user
error or a module/documentation error?
Thanks,
Callum
--
Callum
s(string,null)
> description(string,null)
> >
> > 1:200:1:bin\:192.168.99.201\:60200:1:3:50:NULL:NULL:node-a
> > 2:200:2:bin\:192.168.99.202\:60200:1:3:50:NULL:NULL:node-b
> >
> >
> > Cheers,
> >
> > Fabian
> >
> > - Ursprüngliche Mail ---
32 opensips-test-mtl /usr/local/sbin/opensips[66656]:
> INFO:tls_mgm:init_ssl_ctx_behavior: client verification NOT activated.
> Weaker security.
> Sep 04 13:51:32 opensips-test-mtl /usr/local/sbin/opensips[66656]:
> ERROR:tls_mgm:load_certificate: unable to load certifica
Hi OpenSIPs Community,
I wanted to report an issue I discovered when attempting to use the
rtpproxy module, using the dialog backed rtpproxy_engage() function.
Finding that it was not engaging in certain scenarios I took a closer look
at the activity on the control port and discovered that OpenSIP
5.813619 127.0.0.1:58127 -> 127.0.0.1:4545
> 3795_11 Lc0,3,110,8,98,101 91f0ad91-bde974e4@10.10.0.12 10.10.0.15 8000
> d6119
> dad8d81e378o2;1 5a60fb3a;1
>
> Am I missing something here ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>
Hi All,
I am working on a problem where for a few destinations my OpenSIPs is
receiving RE-INVITE messages with late SDP. This is causing a breakdown in
the rtpproxy engagement and causing the audio to fail mid call.
The OpenSIPs deployment is acting as a SIP proxy which traverses NAT and
rtpprox
r/answer model.
>
> [1] https://github.com/OpenSIPS/opensips/issues
>
> Best regards,
> Răzvan
>
> On 5/15/19 7:36 PM, Callum Guy wrote:
> > Hi All,
> >
> > I am working on a problem where for a few destinations my OpenSIPs is
> > receiving RE-INVITE messages
https://github.com/OpenSIPS/opensips/issues/1702
On Mon, 20 May 2019 at 13:49, Callum Guy wrote:
> Hi Răzvan,
>
> Very kind of you to review this problem. I will go ahead and raise an
> issue as requested.
>
> I can confirm that I have had some success using the offer/ans
Hi Mark,
Can you confirm which engagement functions you are using? Does the SDP look
like its being handled for the RE-INVITE transactions?
I have dealt with some similar scenarios recently and would highly
recommend using the separate rtpproxy_offer() and rtpproxy_answer() methods
over the dialo
generally recognised method of re-engaging rtpproxy
> when sessions already exist? I'm thinking the 'l' flag is required?
>
> Regards
> Mark.
>
>
> On Tue, 11 Jun 2019 at 09:16, Callum Guy wrote:
>
>> Hi Mark,
>>
>> Can you confirm which engagement
Hi All,
I am in the process of configuring a WebRTC/SIP proxy and wanted to know if
there was any way to prevent OpenSIPs from attempting to resolve the
.invalid domains present in the Via headers:
INVITE sip:a...@us.net SIP/2.0
Via: SIP/2.0/WSS b7evbq9f9q22.invalid;branch=z9hG4bK5495466
Althoug
> https://www.opensips.org/events/Summit-2019Amsterdam/
>
> On 06/12/2019 12:08 PM, Callum Guy wrote:
>
> Hi All,
>
> I am in the process of configuring a WebRTC/SIP proxy and wanted to know
> if there was any way to prevent OpenSIPs from attempting to resolve the
> .inval
You might find that a tcpdump is the only way to get to grips with the
underlying issue.
Having said that I wonder if there is any chance that the connection isn't
accepting simply due to a cipher incompatibility. Are you setting a cipher
list that you know your clients accept? Maybe try:
modpara
Hi David,
This seems to be a database collation issue - did you create the database
manually and do you have the option to change the charset?
https://stackoverflow.com/questions/1814532/1071-specified-key-was-too-long-max-key-length-is-767-bytes
Not sure what the correct options would be for yo
Hi All,
My config integrates with an external routing API such that the call ID,
source and URI data (etc) are all provided to a service in URL parameters
via rest_get. The service returns some routing information such as revised
target URI and other options.
To improve this service and protect a
Hi Solarmon,
I can't comment on the exact behaviour internally regarding the ticks value
however I thought I could share it as I see it from a user perspective.
The relevant settings I use are as follows:
modparam("rtpproxy", "rtpproxy_disable_tout", 60)
modparam("rtpproxy", "rtpproxy_timeout",
I think your syntax is wrong?
https://opensips.org/html/docs/modules/3.0.x/sipmsgops.html#func_add_body_part
On Fri, 27 Sep 2019 at 08:36, Sasmita Panda wrote:
> How I will add a attribute in the SDP body . In the rtpengine module there
> is no flag which will add a attribute directly .
>
> If
Hopefully just in prep for 3.0.1!
On Mon, 30 Sep 2019 at 08:42, Adrien Martin wrote:
> Hello,
>
>
> Both apt.opensips.org and yum.opensips.org seem down (IPv4 and IPv6).
>
>
> Regards,
>
> Adrien Martin
>
> ___
> Users mailing list
> Users@lists.opensi
Hi All,
I wanted to report the following behaviour in 3.0 - launch wrapping
rest_get does not continue execution:
route {
xlog("L_INFO", "INFO: LAUNCHING.! \n");
rest_get("https://www.google.com";, $avp(response_body));
xlog("L_INFO", "INFO: LAUNCHED. \n");
}
2019-10-01T09:25:5
Hi All,
Hopefully a simple question related to the behaviour of the presence module
on my registrar.
I have configured OpenSIPs to enable dialoginfo updates against specific
user agents. When the user agent places/receives a call the PUBLISH event
is generated correctly. This resolves a UA at a l
Hi All,
Using 3.0.1
I'm dealing with a call scenario where a UA is dialling out using TLS and
travelling between two geographic sites, both of which use NAT. TLS is only
required between the UA and the initial registrar, once the initial
requests hit the network it is forwarded using cleartext UD
The answer is in your email - the device and server don't share any ciphers.
You'll have to find an acceptable cipher which meets your security
requirements and which your phone supports. I'd suggest you either find a
manual for the phone or run a trace to look into the handshake or similar
If yo
Hi All,
I've been testing a new cluster based registrar config and noticed that the
location records weren't being written back to the database.
The issue appears resolved when I switch usrloc.restart_persistency from
"sync-from-cluster" to "load-from-sql". This is acceptable however it seems
mor
the code. It seems to be still functional,
> so as
> long as you enable it, the DB writes will be denied.
>
> Regards,
>
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 06.11.2019 17:10, Callum Guy wrote:
>
> Hi All,
>
Cheers,
>
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 06.11.2019 18:52, Callum Guy wrote:
>
> Thanks Liviu, that helps a lot. My expectation was that
> the sync-from-cluster option would favour a sync operation from a local
> active peer v
New installation running on OpenSIPs 3.0.1.
Quick question, has anything changed with regards to the contact_id field
of the location tables? The platform I am using implements the database on
MySQL NDBCLUSTER however the autoincrement values are going wild - I'm
trying to work out if its my datab
.
Callum
On Fri, 8 Nov 2019 at 17:13, Callum Guy wrote:
> New installation running on OpenSIPs 3.0.1.
>
> Quick question, has anything changed with regards to the contact_id field
> of the location tables? The platform I am using implements the database on
> MySQL NDBCLUS
You might want to read up on bridge mode, it allows you to meet the finely
control which interface is presented during the SDP rewrites.
All of the information on the various use cases is available in the module
docs, I've used both successfully including some pretty complex request
routing.
The
l routing and again in the failure route breaks the SDP
> which I believe is expected.
> If I don't call rtpproxy_offer for the initial INVITE then the SDP is
> broken for that leg.
>
> Clearly I'm missing something somewhere...
>
>
>
> On Tue, 19 Nov 2019 at 13:28, Ca
Hi All,
I have recently deployed a new registrar and have been seeing a gradual
increase in the memory footprint - enough that I'm having to expand the RAM
(its virtualised) to ensure it doesn't run out.
You can see a diff of the statistics collected last night at 11pm and today
at 3pm here:
http
y need to investigate some of those to find out where your memory is
> going, or look at other processes/daemons you have running that could be
> using that memory.
>
>
>
> Ben Newlin
>
>
>
> *From: *Users on behalf of Callum Guy <
> callum@x-on.co.uk>
rorsi05: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 hel
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 rule
much appreciated, thanks.
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,
>>
&g
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: david.
message was
caught 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 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 wo
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 m
o get rid of the private 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,
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 opt
uchircu | www.opensips-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
ually, 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: *ope
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, R
a
> OpenSIPS 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 Gith
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 cor
: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
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 e
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:
https://www.opensip
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 w
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')
INFO:usrloc:receive_ucon
s 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_
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
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 registrat
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 su
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 be
yours, so it may not be the same.
>
>
>
> [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: *Mo
0) || 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
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 are
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) = $var(rule_attrs);
> .
Hi Sasmita,
I would advise that you capture this information in a branch flag during
registration, these will be stored in location and retrieved when
performing a matching lookup().
So:
if ($pr == "ws" || $pr == "wss") {
setbflag(SRC_WS);
}
save("location_table")
When you do the lookup thi
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:
https://github.com/OpenSIPS/opensips/blob/7dd1151341b8229cd30e335
ei Iancu
>
> 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 confi
sed_size": 12793792,
"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
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) h
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
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
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 wrote
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,
>
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 accordanc
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
> http://
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? 😀
>
> https://www.youtube.com/playlist?list=PL
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:
>
>
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 the
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
>>
>>
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 $(dlg_val(dialog_min_time){s.int})
On Tue, 18 Aug 2020 at 17:35, Igor Pavlov wrote:
>
> Hi all
ot clear how it chooses which one to use for
conditional statements etc
On Thu, 20 Aug 2020 at 07:35, Igor Pavlov wrote:
>
> Thanks a lot for {s.int} ! I forgot that my dialog value is string.
> Transforming to int helped.
>
>
> ср, 19 авг. 2020 г. в 02:33, Callum Guy :
>>
>&g
Hi All,
Noticed these in my logs suggesting that something has jammed in the
async tables:
2020-08-20T15:42:24.145805+01:00 FR-P-SIPSBC-1 opensips[240129]:
WARNING:rest_client:get_multi: max async transfers! (250)
2020-08-20T15:42:24.146290+01:00 FR-P-SIPSBC-1 opensips[240129]:
WARNING:rest_clien
wrote:
>
> On 20.08.2020 17:46, Callum Guy wrote:
> > I presume this is fallout from a recent network issue so I plan to
> > restart all instances during a quiet period which I'm sure will
> > resolve it.
> >
> > Is there any change that visibility of the async
Have you double checked it's not a firewall issue on the SIP proxy?
The transport is typically UDP so there is a fair chance it's blocked.
On Tue, 27 Oct 2020 at 19:05, John Quick wrote:
> Using OpenSIPS v2.4.8 and the latest release of rtpengine. The same was
> also
> happening with some older
I believe the DB followed by a dr_reload is your only option...
On Fri, 30 Oct 2020 at 15:49, Mark Farmer wrote:
> Hi everyone
>
> I am looking for a way to manage routes etc in drouting without using
> opensips-cp.
> I was hoping to find a MI function to add/remove routes but there only
> seems
An alternative option would be to leverage cachedb_local and opensips-cli
to implement your list of accounts and rate limits. It has the advantage of
using the internal opensips cache service and is probably your most high
performance option, with the CLI you can automate data refreshes using
basic
Maybe permissions or similar - have you tested the example from the docs?
exec("ls -l", , "$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
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
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 base
Happy SIPing all.
On Mon, 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.
>
&
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 and
enSIPS 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 t
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 us
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 h
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
>
>
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
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