We have a mid-registrar (OpenSIPS 3.1) in front of an Asterisk node.
Mid-registrar saves with the 'memory only' flag. We are using AOR throttling
We've had a few occasions where something has gone wrong with the
registration on Asterisk so that Asterisk thinks an extension is no longer
.
>
>
> Bottom line, in my opinion, you need to have 2 separate streams before you
> can start STT.
>
>
> wkr,
>
>
> On 17/09/2021 11:04, Mark Allen wrote:
>
> I'm just starting to look at Speech-to-Text (STT) processing for calls -
> initially recordin
I'm just starting to look at Speech-to-Text (STT) processing for calls -
initially recordings but moving on to real-time. I would see this working
along the lines of either:
- a call is recorded, and when the call ends an event is triggered to
initiate transcription of the recording
- a call
'=' and is there to avoid parsing errors in
RTPEngine
On Wed, 15 Sept 2021 at 10:58, Johan De Clercq wrote:
> No worries
>
> On Wed, Sep 15, 2021, 11:43 Mark Allen wrote:
>
>> Ah! Thanks Răzvan and Johan. I was thinking that the options described in
>> the OpenSIPS RTPEng
han suggests, follow the rtpengine documentation, you will
> find more info there.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 9/14/21 15:51, Mark Allen wrote:
> > Hi Răzvan
> >
> > OpenSIP
Hi Răzvan
OpenSIPS passes the commands to RTPengine as you supply them, so in
> theory this should be supported. Did you try to set this up? Can you
> provide any logs about this?
>
>
No. I'm not sure how I would specify this in the opensips.cfg
rtpengine_offer parameters. From the documentation
Hi all,
I'm using RTPEngine for transcoding between Opus and PCMU. According to the
RTPEngine documentation (https://github.com/sipwise/rtpengine), it can
accept formatting parameters for CODECs that support them. For instance,
Opus allows you to change the maxaveragebitrate. However, in the
Ah! Scratch that. I've got it working now. What I was missing was that the
version for apt was 3.1.5-1, so...
apt install opensips=3.1.5-1
...worked for me
On Tue, 24 Aug 2021 at 08:03, Mark Allen wrote:
> Hi all,
>
> We're using a Debian Buster system. I've tried installing 3.
Hi all,
We're using a Debian Buster system. I've tried installing 3.2 but hit
problems with the database migration. I've tried to revert to 3.1 but can't
seem to access that versions package - 3.2 is the only one I can see.
As per https://apt.opensips.org/packages.php?v=3.1 I've entered...
ull/2600
>
> Thanks,
> John Burke
>
>
> On 8/20/21 5:00 AM, Mark Allen wrote:
>
> I've not been able to find the answer to this. Can anyone help?
>
> On Thu, 22 Jul 2021 at 11:02, Mark Allen wrote:
>
>> In the rtpengine documentation [1] in the sec
I've followed the instructions at:
https://www.opensips.org/Documentation/Migration-3-1-0-to-3-2-0 but
OpenSIPS is now erroring because of DB problems. I see errors along the
lines of:
ERROR:core:db_check_table_version: invalid version 0 for table xcap
found, expected 4
If I go into the
I've not been able to find the answer to this. Can anyone help?
On Thu, 22 Jul 2021 at 11:02, Mark Allen wrote:
> In the rtpengine documentation [1] in the section "1.2 - Multiple RTP
> proxy usage" it says...
>
> "The balancing inside a set is done automa
In the rtpengine documentation [1] in the section "1.2 - Multiple RTP proxy
usage" it says...
"The balancing inside a set is done automatically by the module based
on the weight of each RTP proxy from the set."
...how is the weighting determined? Is there a parameter to allocate a
weighting
Hi Liviu
On Wed, 21 Jul 2021 at 16:45, Liviu Chircu wrote:
>
> > ERROR:core:unescape_user: invalid hex digit <37>
> > ERROR:path:path_rr_callback: failed to unescape
> > received=sip:35.x.x.x:60026%%3btransport%%3dtls
>
> ...Which scenario of the two below would you
> say we are in, and with
sk and Asterisk then transcodes, this will
> likely be less efficient.
>
>
> So obviously it's not as simple as saying one will always outperform the
> other, however, there are probably more scenarios in which option 2 would
> be preferable.
>
>
> On 2021-07-19 08:
nonack: sending request failed
On Wed, 21 Jul 2021, 10:56 Liviu Chircu, wrote:
> On 20.07.2021 13:26, Mark Allen wrote:
> > On registration add_path_received() works but the received address is
> > not formatted correctly. I am seeing '%%3b' instead of a semicolon,
> > and
Hmm - not sure what went wrong with my post. It seemed to add some unwanted
formatting. Here's the failing example again...
Path:
Path:
On Tue, 20 Jul 2021 at 10:49, Mark Allen wrote:
> I'm seeing a strange problem when using add_path_received() for
> registration of NATed UAC
I'm seeing a strange problem when using add_path_received() for
registration of NATed UAC using TLS with OpenSIPS 3.1
On registration add_path_received() works but the received address is not
formatted correctly. I am seeing '%%3b' instead of a semicolon, and '%%3d'
instead of an equals sign.
I wonder if anyone can offer any insights...
We are using OpenSIPS 3.1 as a mid-registrar and in front of an Asterisk
box. We include incoming WebRTC traffic using the OPUS codec. Which do you
think would be the better option:
1 - Pass OPUS directly through to Asterisk
2 - Use RTPEngine to
So, we now have a fix for this. Here's a summary of the error we were
seeing and the solution in case it's useful to anybody else.
The problem we were seeing was a SIP message failing to parse and raising
errors. This was particularly problematic for us when it was a REGISTER
failing. Looking at
the entire connection and see where it
> starts breaking.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 5/26/21 2:15 PM, Mark Allen wrote:
> > I'm seeing a weird, intermittent error. The most common occu
Ticket raised - https://github.com/OpenSIPS/opensips/issues/2533
On Fri, 28 May 2021 at 07:32, Mark Allen wrote:
> Thanks Bogdan - will do
>
> On Fri, 28 May 2021 at 06:54, Bogdan-Andrei Iancu
> wrote:
>
>> Hi Mark,
>>
>> Consider opening a ticket on github.
&
PS Bootcamp 2021 online
> https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> On 5/27/21 2:57 PM, Mark Allen wrote:
>
> This seems to link to a past thread where error_route is not triggering if
> the first line is malformed. Unfortunately, there is no resolution to this
> pas
This seems to link to a past thread where error_route is not triggering if
the first line is malformed. Unfortunately, there is no resolution to this
past thread - link here:
https://opensips.org/pipermail/users/2019-February/040605.html
On Thu, 27 May 2021 at 11:56, Mark Allen wrote:
>
>
In my script, I have an error_route...
error_route {
xlog("--- error route class=$(err.class) level=$(err.level)
info=$(err.info) rcode=$(err.rcode) rreason=$(err.rreason) ---\n");
xlog("--- error from [$si:$sp]\n+\n$mb\n\n");
sl_send_reply($err.rcode, "$err.rreason");
I'm seeing a weird, intermittent error. The most common occurrence is with
a 200OK returned by Mid-Registrar on a re-REGISTER using registration
throttling, but we see it elsewhere. It appears that the 200OK message is
getting garbled.
We have a bit of a weird setup to overcome issues we were
p_chan it would be something like:
>
> [opensips]
> type=friend
> deny=0.0.0.0/0.0.0.0
> permit=OPENSIPS_IP/255.255.255.255
> host=OPENSIPS_IP
>
> You definitely can set this up with FreePBX web ui.
>
>
>
> On Mon, Mar 29, 2021 at 10:24 AM Mark Allen wrote:
We have a DID. If an incoming INVITE goes via OpenSIPS, Asterisk returns
'401 Unauthorized' requesting authorization credentials. If we map the DID
direct to Asterisk it doesn't ask for authorization. Our setup is...
DID ---> OpenSIPS 3.1 Mid_registrar ---> Asterisk (FreePBX)
Is there something
ER") {
remove_hf("Expires");
}
}
On Fri, 19 Mar 2021 at 19:48, Ricardo Martinez
wrote:
> Do you have some tip to solve this issue ??.
> Did you use the .conf file or did you have to modify the sources ??
>
> Thanks
> Ricardo
>
> El vie., 19 de marzo de 2021 4:04 p.
cardo.-
>
>
>
> *De:* Users *En nombre de *Mark Allen
> *Enviado el:* miércoles, 3 de febrero de 2021 8:42
> *Para:* OpenSIPS users mailling list
> *Asunto:* [OpenSIPS-Users] OpenSIPS 3.1 - Mid_Registrar AOR throttling &
> FreePBX/Asterisk Expiry problem
>
>
>
> Anything specific concerning/confusing you at this stage?
>
Hi Callum. Yes, there are a few things...
1 - Conceptually, I'm not quite clear on how the initial PN link works. My
understanding from the blog and video is that OpenSIPS receives an initial
register that causes it to send a
:
> sending push before you send the invite solves many many problems.
>
> Op wo 17 mrt. 2021 om 08:54 schreef Mark Allen :
>
>> OK - thanks for that. I'll give it a try
>>
>> On Tue, 16 Mar 2021 at 12:22, Johan De Clercq wrote:
>>
>>> Implement push.
>>&
OK - thanks for that. I'll give it a try
On Tue, 16 Mar 2021 at 12:22, Johan De Clercq wrote:
> Implement push.
>
> Op di 16 mrt. 2021 om 13:15 schreef Mark Allen :
>
>> We are using OpenSIPS 3.1 as a mid_registrar in AOR Throttling mode. When
>> a mobile moves from da
We are using OpenSIPS 3.1 as a mid_registrar in AOR Throttling mode. When a
mobile moves from data to wifi or vice versa, my understanding is that the
registration on OpenSIPS becomes stale (because the phone has a new IP
address). Does this mean that the mobile phone is not registered until
You can also use the "subst" function from the textops module to make
changes to the Contact HF using a regexp
On Fri, 12 Mar 2021 at 14:50, Joseph Barrero wrote:
> Ovidiu, thank you for the suggestion. I don't know why I didn't think
> that remove_hf would work, but after I re-read the module
working fine now. Thanks for the help
On Wed, 10 Mar 2021 at 10:39, Mark Allen wrote:
> Hi Callum - thanks for that!
>
> Yes - it's generating the BYE at the Linux end but not sending it to the
> remote OpenSIPS IP address but rather to an address on the local LAN -
> hence the problem.
e BYE is being generated and sent somewhere else?
>
> Callum
>
> On Tue, 9 Mar 2021 at 16:32, Mark Allen wrote:
>
>> I'm seeing some odd behaviour which also leads into a broader question
>>
>> I have a NATed Blink app running on Linux on my home LAN. It connects to
>
I'm seeing some odd behaviour which also leads into a broader question
I have a NATed Blink app running on Linux on my home LAN. It connects to an
OpenSIPS 3.1 server in on our office LAN which is a mid-registrar for an
Asterisk server. I'm running sngrep on the OpenSIPS box to watch the
traffic.
rr)");
On Thu, 4 Feb 2021 at 11:29, Dragomir Haralambiev
wrote:
> Hi,
>
> When try to start Opensips with follow settings:
> exec("ls -l", , $var(out), $var(err), $avp(env));
>
> I receive follow error:
> column 12-21: syntax error
>
> bad arguments
Hi Dragomir,
exec is working for me in OpenSIPS 3.1 with command...
exec("/root/scripts/script.sh $si", , $var(out), $var(err));
Is it the quotes around $var(err) that is causing the null output??
On Thu, 4 Feb 2021 at 10:37, Dragomir Haralambiev
wrote:
> Hello,
>
> I try to run external
I'm seeing strange behaviour using mid_registrar with AOR throttling...
On initial registration, I do a mid_registrar_save():
mid_registrar_save("location","mp0v","sip:$tU@midreg",,"vipx");
Return value from save is "1" (success) and then I successfully forward the
REGISTER to the
Further to this - as I said the relay_ip overcame the immediate audio
problem, but on testing it timed out after just over 60 seconds. Looking at
the traffic in Wireshark and the SDP in SIP messages the cause seems to be
that Asterisk is sending RTP direct to the 46.xxx.xxx.xxx address rather
than
Hi John and Johan - thanks for your replies. I'll have a look at RTPEngine
to see if it makes things simpler for me.
I have managed to get audio working both ways with Mediaproxy - the problem
I was encountering was with config.ini settings. I had to explicitly set
"relay_ip" and restarted
Our setup...
External UAC 192.168.x.x
|
Router5.x.x.x
|
(internet)
|
Firewall 46.x.x.x maps ports
| directly to
OpenSIPS 192.168.x.x Mid-registrar and Mediaproxy
|
Asterisk 192.168.x.x
Current situation:
- UAC can call destination registered on
isk, forwarding the rtp port range configured in
> asterisk from the firewall to asterisk should do it.
>
>
> On Thu, 14 Jan 2021 at 08:23, Mark Allen wrote:
>
>> Thanks Adrian
>>
>> The firewall has SIP-ALG disabled and just forwards ports from externally
>&g
SIP ALG problem to see if this is relevant for your case.
>
> Regards,
> Adrian
>
>
> On 13 Jan 2021, at 13:08, Mark Allen wrote:
>
> Hi all - I've been banging my head against this but not succeeding.
>
> Our setup...
>
> UAC 192.168.x.x
> |
Hi all - I've been banging my head against this but not succeeding.
Our setup...
UAC 192.168.x.x
|
Router5.x.x.x
|
(internet)
|
Firewall 46.x.x.x maps
| directly to
OpenSIPS 192.168.x.x Mid-registrar
|
Asterisk 192.168.x.x
Current
info: $fU, $tU, $td,
$rm");
However, I've had multiple timers kicking off LUA scripts all running for a
while now and I've not seen any problems. YMMV
On Wed, 23 Dec 2020 at 15:17, johan wrote:
> then you are fine :-)
> On 4/12/2020 09:26, Mark Allen wrote:
>
> Interestingly -
Sorry... should have added that OpenSIPS box is acting as mid-registrar
On Thu, 7 Jan 2021, 12:12 Mark Allen, wrote:
> I wonder if anyone can help me with this? I am trying to configure
> Mediaproxy to handle RTP traffic coming from outside our local network.
> Here's the setup:
&g
I wonder if anyone can help me with this? I am trying to configure
Mediaproxy to handle RTP traffic coming from outside our local network.
Here's the setup:
UAC ---> IPA ---> IPB ---> Mediaproxy / OpenSIPS ---> Asterisk
IPA (a public IP address 4x.xxx.xxx.xxx) maps ports ports 5060 and 1
Yep - the software loaded successfully, integrated with OpenSIPS and I can
see in the log file that it is trying to handle RTP traffic. Just need to
work out the correct configuration now! :)
Thanks for the help Adrian
___
Users mailing list
bug.
>
> You must update to the latest mediaproxy version:
>
> sudo apt update
> sudo apt install mediaproxy-relay mediaproxy-common mediaproxy-dispatcher
>
> Regards,
> Adrian
>
> On 6 Jan 2021, at 12:59, Mark Allen wrote:
>
> Hi all - not sure what I'm missing her
Hi all - not sure what I'm missing here...
I'm installing Mediaproxy onto our Debian Buster box which is also running
OpenSIPS 3.1 but when I start the relay I'm getting an error in syslog...
15:40:07 opensipsx media-relay[4983]: INFO Starting MediaProxy Relay
4.0.4
15:40:07 opensipsx
Hi Juan Carlos - I feel your pain!
I've finished for the year so I don't have access to the code at the
moment. I'll have a look at this in the new year to see if I can post
something sensible! :-)
Short version is that it turned out to be a horrible challenge to get it
working as desired. My
, , DUMMY
...so LUA will be passed a (very simple) message it seems
On Thu, 3 Dec 2020 at 15:57, Mark Allen wrote:
> LOL! Yes, I did understand, but it is an important distinction.
>
> On Thu, 3 Dec 2020 at 15:53, Ben Newlin wrote:
>
>> It seems like you read that as I i
wlin
>
>
>
> *From: *Users on behalf of Mark Allen <
> m...@allenclan.co.uk>
> *Date: *Thursday, December 3, 2020 at 10:40 AM
> *To: *OpenSIPS users mailling list
> *Subject: *Re: [OpenSIPS-Users] lua_exec in timer route - OpenSIPS 3.1
>
> > a memory leak or segf
would feel safe
> assuming that this wouldn’t result in a memory leak or segfault after
> continued use.
>
>
>
> Ben Newlin
>
>
>
> *From: *Users on behalf of Mark Allen <
> m...@allenclan.co.uk>
> *Date: *Thursday, December 3, 2020 at 10:04 AM
> *To: *OpenSIPS users
rs] lua_exec in timer route - OpenSIPS 3.1
>
> what you can try, is to call another route in the time route.
>
> And then in that route, you execute the lua script.
>
> maybe (just a myabe) that will work.
>
>
>
> wkr,
>
>
>
> Op do 3 dec. 2020 om 12:23 schree
id you try executing a script in timer route ?
> What's the output in the log ?
>
> Op do 3 dec. 2020 om 11:56 schreef Mark Allen :
>
>> Is there a way to run a lua_exec from a timer route?
>> ___
>> Users mailing list
Is there a way to run a lua_exec from a timer route?
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Good point - I'll do that
On Wed, 2 Dec 2020 at 16:33, Liviu Chircu wrote:
> On 02.12.2020 18:16, Mark Allen wrote:
>
> Thanks so much for your help Liviu. In the end, I used the $shv() approach
> you suggested. It worked like a dream! :-)
>
> In opensips.cfg:
>
>
shv(cacheValid) == 0) {
xlog("Reloading cache");
...code to reload cached data goes here...
$shv(cacheValid) = 1;
}
Trigger reload using opensips-cli:
opensips-cli -x mi shv_set cacheValid int 0
On Wed, 25 Nov 2020 at 14:26, Liviu Chirc
Is there a simple way to test if a cache has any values stored?
I have a cache collection holding a number of IP addresses that are read on
startup from a local DB. If the DB table is updated I can flush the cache
of the old data with "opensips-cli -x mi cache_remove_chunk". I would then
want to
remove_chunk "*" other
>
> Regards,
>
> --
> Vlad Patrascu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 20.11.2020 15:25, Mark Allen wrote:
>
> thanks Vlad - I've tried it but I get
>
> ERROR: command 'cache_remove_chunk' returned: -32602: I
also update the docs as the example is outdated so thanks for
> pointing it out!
>
> Regards,
>
> --
> Vlad Patrascu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 20.11.2020 13:47, Mark Allen wrote:
>
> I have local cache collections setup in my sc
Sorry - accidentally tagged this on to another thread, so please excuse the
repost...
I have local cache collections setup in my script:
loadmodule "cachedb_local.so"
modparam("cachedb_local", "cache_collections", "general; other;")
modparam("cachedb_local", "cachedb_url",
I have local cache collections setup in my script:
loadmodule "cachedb_local.so"
modparam("cachedb_local", "cache_collections", "general; other;")
modparam("cachedb_local", "cachedb_url", "local:general:///general")
modparam("cachedb_local", "cachedb_url", "local:other:///other")
I
.org/docs/modules/3.1.x/cachedb_local.html
> https://opensips.org/docs/modules/3.1.x/sql_cacher.html
> https://opensips.org/docs/modules/3.1.x/avpops.html
>
> On Sun, Nov 15, 2020 at 9:49 AM Mark Allen wrote:
> >
> > Being a bit dense here - the documentation says to set mo
t;> On 13/11/2020 16:06, Ovidiu Sas wrote:
>>
>> Take a look at db_text and sql_cacher modules!
>>
>> Regards,
>> Ovidiu Sas
>>
>> On Fri, Nov 13, 2020 at 09:50 Mark Allen wrote:
>>
>>> Just would like to consult the hive mind. I want to
hon? Parsing is trivial and you can call internal function i.e.
> cache_store()
> right from your Python code.
>
> -Max
>
> On Fri., Nov. 13, 2020, 6:50 a.m. Mark Allen,
> wrote:
>
>> Just would like to consult the hive mind. I want to read the contents of
>> a multi-l
Just would like to consult the hive mind. I want to read the contents of a
multi-line text file to be used by my OpenSIPS config. Ideally, I'll get a
key:value CSV pair from the file and store each pair in memcache - e.g.
file contains:
a, 113
b, 214
c, 771
read it in line by line and
+--+
>
> If you do not have "path" set in your case the problem is probably there.
> My lookup is not mid_register but it is close to what you have. I on
d_register() and
fix_nated_contact() but it made no difference.
On Fri, 21 Aug 2020, 13:23 Slava Bendersky via Users, <
users@lists.opensips.org> wrote:
> Please check contact header.
>
> volga629
>
> ------
> *From: *"Mark Allen"
> *To: *"OpenSI
ug 2020 at 08:44, Mark Allen wrote:
> I don't know if anyone has had a chance to look at my problem but I wonder
> if at least I could get an opinion on the following:
>
> 1 - Should I be seeing the path saved in the appropriate column in the
> "location" tabl
Thanks Johan - I'll try this out
On Mon, 3 Aug 2020 at 11:25, Johan De Clercq wrote:
> t_relay to the socket on which you are listening.
>
> Op ma 3 aug. 2020 om 12:21 schreef Mark Allen :
>
>> > If you want to see it, loopback the message.
>>
>> Thanks. How
> If you want to see it, loopback the message.
Thanks. How do I do that?
On Mon, 3 Aug 2020 at 11:02, Johan De Clercq wrote:
> I think that you are right.
> If you want to see it, loopback the message.
>
> Op ma 3 aug. 2020 om 11:16 schreef Mark Allen :
>
>> Woul
entation/Development-Manual
On Thu, 30 Jul 2020 at 16:24, Mark Allen wrote:
> Seeking to find a workaround for the problem I'm having with WebRTC and
> AOR throttling (
> http://lists.opensips.org/pipermail/users/2020-July/043542.html) I want
> to access the values of the "V
uot;location")) {
t_reply(404, "Not Found");
exit;
}
NB - route(resolve_registrar) sets the variable $avp(main_registrar) to the
IP address of the Asterisk server
On Thu, 30 Jul 2020 at 09:16, Mark Allen wrote:
> We are working on a test setup, hoping to move to a production
Seeking to find a workaround for the problem I'm having with WebRTC and AOR
throttling (http://lists.opensips.org/pipermail/users/2020-July/043542.html)
I want to access the values of the "Via" and "Path" headers that are being
passed to the registrar.
Using sngrep on the OpenSIPS server I can
We are working on a test setup, hoping to move to a production system in
mid-August. We want to use mid-registrar AOR throttling. Users will connect
through OpenSIPS using a combination of SIP and WebRTC endpoints,
registering to an extension on an Asterisk main-registrar...
ately causing a crash.
>
> Regards,
>
> --
> Vlad Patrascu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 28.07.2020 17:42, Mark Allen wrote:
>
> [SOLVED]
>
> Hi Stas - good call! It's a change in behaviour from 3.0.
>
> In 3.0 documentation sa
:)
On Tue, 28 Jul 2020 at 15:50, Stas Kobzar wrote:
> I mean, you are welcome, Mark :) sorry
>
> On Tue, Jul 28, 2020 at 10:45 AM Mark Allen wrote:
>
>> [SOLVED]
>>
>> Hi Stas - good call! It's a change in behaviour from 3.0.
>>
>> In 3.0 documentation
xlog("Raised E_WFC_REGISTERED $avp(values)");
> raise_event("E_WFC_REGISTERED", *$avp(keys)*, $avp(values));
>
> I know they are said to be optional in the documentation but probably it
> is optional for two. Either no params or if you pass parameters, you
We're upgrading from 3.0 to 3.1. Everything seems ok except we get a weird
error. We subscribe a dynamic event...
startup_route {
subscribe_event("E_WFC_REGISTERED", "udp:127.0.0.1:");
}
which we can see works from /var/log/syslog...
event_datagram:mod_init: initializing
Thanks Liviu. That worked.
On Tue, 28 Jul 2020 at 09:12, Liviu Chircu wrote:
> On 28.07.2020 11:00, Mark Allen wrote:
> > Presumably, there's somewhere that I should be telling opensips-cli
> > that I want it to use the 3.1 schema?
>
> Hey, Mark!
>
> That seems to
3.0_to_3.1 opensips opensips_mig_3_1
...which seems to have upgraded to the correct version
On Tue, 28 Jul 2020 at 09:00, Mark Allen wrote:
> Presumably, there's somewhere that I should be telling opensips-cli that I
> want it to use the 3.1 schema?
>
> On Tue, 28 Jul 2020 at 08:54
Presumably, there's somewhere that I should be telling opensips-cli that I
want it to use the 3.1 schema?
On Tue, 28 Jul 2020 at 08:54, Mark Allen wrote:
> Getting version table errors on startup after move to 3.1...
>
> ERROR:core:db_check_table_version: invalid version 10 for tab
Getting version table errors on startup after move to 3.1...
ERROR:core:db_check_table_version: invalid version 10 for table dialog
found, expected 11
Version 3.1 is built from 3.1 branch on GitHub. I think I've followed the
documentation correctly but OpenSIPS 3.1 is not starting and I'm
c opensips use path module and function add_path_received
>
> On Tue, Jul 14, 2020 at 11:14 AM Mark Allen wrote:
> >
> > I'm new to OpenSIPS and I've hit a problem I can't find a way past
> >
> > We have a test setup with an OpenSIPS mid-registrar in front of an
> A
; lack of resources, this hasn't reached our priority list.
> Nevertheless, that is a wiki page, and any contribution is more than
> welcome.
>
> [1] https://www.opensips.org/Documentation/Tutorials#toc9
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
>
Hi Alexey - thanks for responding.
I've seen past reports where NAT was causing this type of problem. I tried
your suggestion but, along with other tests such as forcing
fix_nated_register() or fix_nated_contact() on all messages, and after
trying Stas' suggestion, it still doesn't work for me. I
Not sure where to report this, so apologies if it's in the wrong place.
The tutorial for Web Sockets with 3.0 looks to be wrong when running 3.0.2.
Example script is full of obsolete commands, modules and variables - not
very helpful. Will this be rectified on release of 3.1 Stable?
this problem.
> https://opensips.org/docs/modules/3.2.x/path.html
>
> In your mid-registerer you need to enable path support. See "save"
> function params p0 and v.
> in your webrtc opensips use path module and function add_path_received
>
> On Tue, Jul 14, 2020 at 11
I'm new to OpenSIPS and I've hit a problem I can't find a way past
We have a test setup with an OpenSIPS mid-registrar in front of an Asterisk
PBX. Mid-registrar is currently in mode 1 (registration throttling). We
have SIP and WebRTC endpoints that we want to use.
*Current state is:*
REGISTER:
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