What memory management and allocator are you using?
I had similar issues using opensips 3.2.7 and the quick workaround was to
change from F_MALLOC to HP_MALLOC. But I thought there should have been
fixes for such issues in newer versions, so your issue might be different.
On Thu, 9 Nov 2023 at
Hi Giovanni,
I personally use PHP to parse the JSON formatted data - but most scripting
and programming languages will have a way of doing it.
I would assume you could use the opensipsctrl (or opensips-cli) and MI JSON
commands to get a script to control opensips - so it will be dependent on
Hi Gionanni,
Thank you for the pointer. In our opensips config, I see that we currently
use:
create_dialog("B")
Cheers!
On Fri, 17 Dec 2021 at 15:12, Giovanni Maruzzelli wrote:
>
> On Tue, Dec 7, 2021 at 8:49 AM solarmon wrote:
>
>>
>> I'm not sure what
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> On 12/14/21 11:05 AM, solarmon wrote:
>
> Hi,
>
> In opensips 2.4.x is there a way to inject a custom field into the dialog
> list? For example, inject a custom SIP header into it?
>
> Thank you!
>
> __
Hi,
In opensips 2.4.x is there a way to inject a custom field into the dialog
list? For example, inject a custom SIP header into it?
Thank you!
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Mon, 6 Dec 2021 at 12:05, Răzvan Crainea wrote:
> Those fields are usually used for re-INVITE pinging. Are you using this
> feature?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 12/2/21 14:45, solarmon
Hi,
In opensips 2.4.x the dialog list contains the following fields for SDP:
* caller_sdp
* callee_sdp
However, these are always empty.
What are these fields used for and how can I get it populated - with, I
assume, SDP information for the dialog?
Cheers!
An alternative method is to use m4 to feed in the detected IP address into
a templated version of the opensips config file.
On Wed, 17 Nov 2021, 08:23 Muhamad Putra Abdullah,
wrote:
> Hi,
>
>
>
> socket=udp:enp1s0:5060 tag INTERNAL_IP # CUSTOMIZE ME
>
> socket=udp:enp2s0:5060 tag EXTERNAL_IP
Hi,
I'm using "opensipsctl fifo dlg_list_ctx" to get information on what active
dialogs there are.
I am also looking for information about what dispatcher and rtpproxy group
the call dialog is using. Is it possible to get this information for the
call dialog, or inject that information into the
lusterer would make set
> it to "backup" on other node.
>
> Cheers,
> Kingsley.
>
>
> On Tue, 2021-07-13 at 14:08 +0100, solarmon wrote:
> > Hi Liviu,
> >
> > I took and used the 'recommended' config as advised at
> >
> https://blog.opensips.or
you.
On Tue, 13 Jul 2021 at 12:59, Liviu Chircu wrote:
> On 25.06.2021 15:12, solarmon wrote:
>
> The typical recommended configuration for an active/standby setup would be:
>
> node1:
>
> modparam("dialog", "dlg_sharing_tag", "vip=active")
&
make sure that a version mismatch not causing the issue.
Thank you.
On Tue, 13 Jul 2021 at 12:50, Liviu Chircu wrote:
> On 25.06.2021 13:26, solarmon wrote:
> > Can anybody confirm whether this behaviour is by design and/or whether
> > how to force the opensips to sync all the Dial
Hi,
Using opensips 2.4.x
I'm testing some call failure scenarios and this particular test is to drop
the IP connectivity to the dispatcher endpoints and see what happens when a
new call is made and before the endpoint is considered unhealthy.
The expected behaviour is that the INVITE will not
Hi,
In a two node cluster with dialog sharing tags configure, the typical
recommended configuration for an active/standby setup would be:
node1:
modparam("dialog", "dlg_sharing_tag", "vip=active")
node2:
modparam("dialog", "dlg_sharing_tag", "vip=backup")
Now imagine if node2 becomes the
Hi,
I am observing an issue with the Dialog syncing between two opensips nodes,
and a corosync/pacemaker cluster setup.
I'm using 'opensipsctl fifo dlg_list' - more specifically 'opensipsctl fifo
dlg_list | grep caller_contact' - to check the Dialogs on both the 'active'
(one with the floating
Hi,
If I had a $fn (From display name) core variable in the format:
"ABC123456789"
I know how to match on "ABC", but how can I extract "1234567879" to put
into another variable?
Thank you.
___
Users mailing list
Users@lists.opensips.org
ike sip msg) ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> OpenSIPS Bootcamp 2021 online
> https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> On 4/16/21 1:28 PM, solarmon wrote:
>
>
nSIPS_eBootcamp_2021/
>
> On 4/16/21 12:22 PM, solarmon wrote:
>
> Hi,
>
> We have a code that response back with a "503 Service not available" if it
> has exhausted all the dispatcher endpoints - see end of email.
>
> It would be useful if we casn send back th
Hi,
We have a code that response back with a "503 Service not available" if it
has exhausted all the dispatcher endpoints - see end of email.
It would be useful if we casn send back the Reason header of the last
rejection. Could this be done and what should I look at?
Thank you!
is reloaded? Can this be done without impacting active
sessions/dialogs?
Thanks.
On Mon, 1 Mar 2021 at 16:11, Liviu Chircu wrote:
> On 01.03.2021 18:04, solarmon wrote:
> > I would like somebody to confirm this is the behaviour - i.e.
> > ds_is_in_list() does a DNS lookup on the Dest
Hi,
Can somebody confirm the following behaviour in opensips 2.4.7
I have a dispatcher endpoint with a DNS address as the Destination address.
When doing a ds_is_in_list() check and passing it the $si (Source IP
address) will it perform a DNS resolution before trying to match the IP?
I see in
, Ovidiu Sas wrote:
> 100rel in the initial INVITE means that the caller has support for it, if
> requested by the caller. If not present in INVITE, the caller should not
> send provisional replies with 100rel.
>
> -ovidiu
>
> On Mon, Jan 18, 2021 at 15:13 solarmon wrote:
>
nge the 183 into an 180 and let the prack take its course.
>
> Alternatively, you can remove 100rel from the initial INVITE and drop the
> 183 without SDP.
>
> -ovidiu
>
> On Mon, Jan 18, 2021 at 11:05 solarmon wrote:
>
>> Hi,,
>>
>> I have a requireme
, wrote:
> Hi,
>
> I guess does not answer the guestion but take a look at this:
>
> https://github.com/OpenSIPS/opensips/issues/2076
>
> that said could you setup b2b logic to fix interworking issues?
>
> Tomi
>
> On 18. Jan 2021, at 18.31, solarmon wrote:
&g
. 2021 om 17:07 schreef solarmon :
>
>> Hi,,
>>
>> I have a requirement to stop 183 without SDP packets from being passed,
>> as well as having to reply back with a PRACK.
>>
>> I can stop the 183 without SDP from being passed on with the following in
>&g
Hi,,
I have a requirement to stop 183 without SDP packets from being passed, as
well as having to reply back with a PRACK.
I can stop the 183 without SDP from being passed on with the following in
onreply_route[]
if ($rs == "183" && !has_body_part("application/sdp")) {
drop();
s/modules/2.4.x/sipmsgops.html#func_codec_exists
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> OpenSIPS Bootcamp 2020 online
> https://opensips.org/training/OpenSIPS_eBootcamp_2020/
>
> On 1/7/21 6:33 P
Hi,
On opensips 2.4.x how would I best check what codec is being offered, and
reject the call if it ONLY offers a codec that is not supported by us. For
example, if we only want to support G.711 PCMA/PCMU.
Thank you.
___
Users mailing list
carriage returns. I only
needed a single '\n' at the start of the replacement text.
It would have been nice for a function to append the end end of the SDP
attributes, but I'm hoping this is a good enough workaround for now.
Thank you once again!
On Tue, 8 Dec 2020 at 15:25, Donat Zenichev
wrote:
Hi,
How can I add/append a ptime value in the SDP packet?
I've tried using:
add_body_part("a=ptime:20", "application/sdp");
However, this just creates an additional "application/sdp" section rather
than add/append to the existing one.
Is there not a sipmsgops or textops function to process
Hi,
As per NICC N1035 document section 6.2 the SIP URI must end with
'user=phone'.
How can this be added to a SIP URI that doesn't contain it?
Is using the add_uri_param() function (
https://opensips.org/html/docs/modules/2.4.x/uri.html#func_add_uri_param)
the best way to do this - i.e.?
OP was that I was using TCP for the
first hep_id Homer server - even though the OP showed I had it for both, my
actual config only had TCP for the first entry.
A previous mailing list email (and subsequent github ticket) from me about
getting Homer 7 to work resulted in showing that it was only worked wit
e uri's and the HEP traffic will only ever go to
the second/last uri defined.
So I know that HEP traffic can get each to the destinations - in terms of
routing and access control.
Why is siptrace not sending to both HEP destinations at the same time?
On Fri, 17 Jul 2020 at 16:49, solarmon wrote:
&
Hi,
I was able to get siptrace sent to a single HEP server. Now I would like to
be able to send to a secondary server as an additional destination.
I thought I could just create additional hep_id and trace_id entries and
run sip_trace() twice, but this didn't work and only the second HEP server
Hi Igor,
Have you tried using the ds_set_state MI command?
(for 2.4.x)
https://opensips.org/html/docs/modules/2.4.x/dispatcher.html#mi_ds_set_state
On Sun, 28 Jun 2020 at 11:35, Igor Pavlov wrote:
> Hi all,
>
> Is it possible to disable ping for particular entry in set at all?
>
> I have a
fferently not with the
>
> $ opensipsctl fifo rtpproxy_show
>
> command
>
> I think you can get the ’status’ directly from the database with SQL
> query.
>
> Tomi
>
>
> On 24. Jun 2020, at 16.33, solarmon wrote:
>
>
> Hi Bogdan-Andrei,
>
> There
SQL table and the `memory
> status` is the status provided by the MU rtpproxy_show.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
>
> On 6/24/20 12:16 PM, solarmon wrote:
>
> Hi,
>
> The comma
Hi,
The command "opensipsctl fifo rtpproxy_show" does not return the 'status'
of the rtpproxy node.
In OpenSIPS Control Panel, the RTPProxy table has a '*Memory State*' column
which seems to be the 'status' of that node.
How can I get this 'Memory State' 'status' in command line form so that it
gt; *Subject: *Re: [OpenSIPS-Users] 502 Bad Gateway events leads to calls
> being rejected with 480 Temporarily Unavailable
>
>
>
> Hello Solarmon,
>
>
>
> I think, The * ds_mark_dst("p");* put your destinations on Probing and
> after a few seconds you will get the re
, wrote:
> Hi,
>
> What is the exact OpenSIPS version you have (opensips -V) ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
>
> On 6/18/20 12:13 PM, solarmon wrote:
>
> Hi,
>
> I'm
Hi,
I'm trying to set up siptrace to send HEPv3 packets to a Homer 7 setup.
Currently this is not working and it seems the HEPv3 packets coming from
opensips is not being ingested for some reason. Using hepgen (
https://github.com/sipcapture/hepgen.js) to generate test HEPv3 traffic on
the same
Hi,
I'm trying to set up and integrate Homer with my opensips setup.
I have managed to install Homer on separate VM, using the docker method as
described at:
https://github.com/sipcapture/homer/wiki/Quick-Install#-quick-install
(I had to set SELINUX to permissive or disable for it to build -
be achieved in opensips when we
respond back to our platform?
Thank you.
On Mon, 8 Jun 2020 at 13:46, Ben Newlin wrote:
> Solarmon,
>
>
>
> Yes, that code is your issue. While it makes sense to mark the remote
> destination unreachable on some error responses possibly, or c
}
# if call failure, allow the reply to propagate to caller
exit;
}
Thank you for the tip about the 'modparam("dispatcher",
"options_reply_codes", "502")' option. I will try that if it is not
recommend to change the above code.
Thank
Hi,
I'm trying to understand whether this is the correct or expected behaviour.
We have two destinations configured in Dispatcher.
What I am noticing is that when we receive 502 Bad Gateway messages (logged
as ("call failed to established with 502 code") from both endpoints. After
both
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 3/27/20 4:43 PM, solarmon wrote:
> > Hi,
> >
> > I have an opensips 2.4 cluster using rtpproxy nodes.
> >
> > I'm troubleshooting some DTMF
Hi,
I have an opensips 2.4 cluster using rtpproxy nodes.
I'm troubleshooting some DTMF issue and have a requirement to change the
inbound INVITE SDP payload from:
a=rtpmap:96 telephone-event/8000
to
a=rtpmap:101 telephone-event/8000
in the outbound INVITE SDP.
How could this
g the TM module and creating
> transactions for all of the 5xx responses, as far as I know.
>
>
>
> Ben Newlin
>
>
>
> *From: *Users on behalf of solarmon <
> solar...@one-n.co.uk>
> *Reply-To: *OpenSIPS users mailling list
> *Date: *Friday, March 20, 2020 at
ssages. XXX_transactions count transaction handled by the TM module.
>
>
>
> So if replies are incrementing but not transactions, it means you are not
> creating transactions for these messages.
>
>
>
> Ben Newlin
>
>
>
> *From: *Users on behalf of solarmon <
sterdam, May 2020
> https://www.opensips.org/events/Summit-2020Amsterdam/
>
> On 3/20/20 1:15 PM, solarmon wrote:
>
> Hi,
>
> I meant that the 5xx_transactions counter value does not increment.
>
> However, I have now found the stats counter "5xx_transactions".
h replies.
>
> 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/
>
> On 3/19/20 2:56 PM, solarmon
Hi,
I'm trying to graph the 4xx and 5xx stats on my opensips cluster. I'm
making use of the "opensipsctl fifo get_statistics" command to extract the
4xx and 5xx replies counters.
The 4xx counter (opensipsctl fifo get_statistics 4xx_replies) seems to
incrementing as expected.
However, the 5xx
:
> On 21.01.2020 15:01, solarmon wrote:
> > Can the drain_mode code just be put at the very start of the route {}
> > block?
>
> If you don't mind letting SIP scanners know when you're doing
> maintenance, sure.
>
> --
> Liviu Chircu
> www.twitter.com/livi
Very good point! I'll try to figure out a more suitable place for it.
Thanks so much for your help!
On Tue, 21 Jan 2020 at 13:06, Liviu Chircu wrote:
> On 21.01.2020 15:01, solarmon wrote:
> > Can the drain_mode code just be put at the very start of the route {}
> > block?
&
Hi Liviu,
Can the drain_mode code just be put at the very start of the route {} block?
Thank you
On Tue, 21 Jan 2020 at 12:29, Liviu Chircu wrote:
> On 21.01.2020 14:21, solarmon wrote:
> > However, now I have checked our opensips.cfg script (that was created
> > for us) it lo
;
}
}
On Tue, 21 Jan 2020 at 11:58, Liviu Chircu wrote:
> On 21.01.2020 13:47, solarmon wrote:
> >
> > So to be clear, I cannot use the dispatcher endpoint method to stop
> > responding to SIP Options pings? If I can do that, then that is the
> > equivalent -
as unhealthy and not
send calls to it.
Thank you.
On Tue, 21 Jan 2020 at 11:43, Liviu Chircu wrote:
> On 21.01.2020 13:06, solarmon wrote:
>
> Thanks for the tip about the $shv global variables. The link you gave is
> for 3.1.x, I found the equivalent for 2.4.x which is the version I'm
' mode?
When I try the 'opensipsctl fifo shv_get debug' on my opensips server I get
the following error:
500 command 'shv_get' not available
I assume I need to have the 'cfgutils' module enabled and loaded for it to
work?
Thank you.
On Mon, 20 Jan 2020 at 15:03, Liviu Chircu wrote:
> Hi solar
Hi,
I have an opensips two node cluster, and using the dispatcher module for
'internal' and 'external' endpoints.
What is the recommended graceful method to put this opensips cluster in
maintenance so that current calls are not affected. The overall effect I am
looking for is for the opensips
2020 at 11:55, Liviu Chircu wrote:
> Hi solarmon,
>
> Please find an elaborate discussion on this topic here [1]. In short,
> MySQL's "wait_timeout" setting directly affects the number of such errors
> you are seeing in the logs, unless you are dealing with some other kin
be
no networking or routing or firewall that could be causing connectivity
issues.
Why is opensips generating such logs suggesting that it cannot connect to
its local database? What could be causing this issue - opensips or the
(MariaDB) database?
Thank you.
On Mon, 23 Dec 2019 at 14:49, solarmon
point in probing mode comes from the SIP
> signaling level - you should add more debugging in your script to
> understand this problem, or take a look at the signaling level.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
Hi,
I'm investigating why there was a blip in the Dispatcher endpoint SIP
Options pings. The endpoint went in to "State=2 (Probing)" state and at the
same time the following was logged in opensips.log:
/usr/local/sbin/opensips[29087]:
INFO:db_mysql:switch_state_to_disconnected: disconnect event
Hi,
A question about the opensipsctl dispatcher show and dump commands. They
seem to show different 'state' for each endpoint.
For example:
When using "opensipsctl dispatcher show":
opensipsctl fifo commands without it hanging. Note, I also had to
restart the NCPA listener service for Nagios monitoring to work properly
again.
I still would like to understand how it got into this situation in the
first place.
On Tue, 17 Dec 2019 at 10:09, solarmon wrote:
> Hi,
>
&g
Hi,
I have an issue with opensipsctl command hanging when using fifo commands.
This occurred after I issued a 'systemctl reload opensips' command.
Actually, I subsequently had to issue a 'systemctl restart opensips'
command since the opensips service had stopped working.
Since then, when using
>From my experience with testing SIPREC on OpenSIPs, I believe the Oreka
solution is not SIPREC compliant - it supports passive and active recording
but does not support the SIPREC protocol.
So maybe, if I'm correct, then this is part of the issue you are having?
I ended up using Drachtio SIPREC
erm "processed" should be read more generically in this context to
> include to act of receiving the replicated dialog and preparing it to
> become active at any time.
>
> Vlad Patrascu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 10/21/2019 02:5
Hi Vlad,
That was my understanding too, but why would processed_dialogs be
incrementing on the standby (State=0) node?
On Mon, 21 Oct 2019 at 12:51, Vlad Patrascu wrote:
> Hi solarmon,
>
> 'failed_dialogs' and 'processed_dialogs' represent the total number of
> failed/proce
corosync/pacemaker for the virtual IP,
and using 'clusterer' between the two OpenSIPS nodes).
On Mon, 21 Oct 2019 at 11:24, Alexey Kazantsev via Users <
users@lists.opensips.org> wrote:
> Hi, solarmon
>
> no, there’s no need to sync dialogs. It’s another type of cluster.
>
>
Hi,
Are dialogs stats meant to be totally synced between to openSIPS nodes in a
cluster - i.e. when using clusterer?
I am monitoring/graphing the active_dialogs, failed_dialogs and
processed_dialogs for each node, using 'opensipsctl fifo get_statistics'
and they are not fully synced:
*Node 1:*
Hi,
If found the solution to my question. I used remove_body_part():
remove_body_part("application/sdp");
Thanks you.
On Fri, 18 Oct 2019 at 09:08, solarmon wrote:
> Hi,
>
> I am able to map 183 to 180 using the change_reply_status() function.
>
> However, I wo
gt;
>>
>> *From: *Users on behalf of David
>> Villasmil
>> *Reply-To: *OpenSIPS users mailling list
>> *Date: *Wednesday, October 9, 2019 at 5:42 AM
>> *To: *OpenSIPS users mailling list
>> *Subject: *Re: [OpenSIPS-Users] Convert 183 to 180?
>&
our system.
Will this change_reply_status() function allow us to do that?
On Mon, 7 Oct 2019 at 14:23, solarmon wrote:
> Hi,
>
> Thanks. I forgot to say that I'm using openSIPS 2.4.x, but the same
> function is available for it too:
>
>
> https://opensips.org/html/docs/modules/2.4.x/sipmsgops.html#c
//opensips.org/html/docs/modules/3.0.x/sipmsgops.html#change_reply_status
>
>
>
> Ben Newlin
>
>
>
> *From: *Users on behalf of solarmon <
> solar...@one-n.co.uk>
> *Reply-To: *OpenSIPS users mailling list
> *Date: *Monday, October 7, 2019 at 5:15 AM
>
Hi,
Our system connected to openSIPS is not handling 183 (with and without SDP)
very well and is non-trivial for us to try to change the behavior. If we
receive an initial 183 without SDP, it does not processed a subsequent 183
with SDP.
We would like to explore how 183 (with and without SDP)
.
On Tue, 20 Aug 2019 at 09:16, solarmon wrote:
> Hi Razvan,
>
> I'm using opensips 2.4.6 (x86_64/linux) so I don't think opensips-cli is
> available?
>
> I'm using opensipsctl to show the rtpproxy status.
>
> This is the output of the command after I have turned off
926858.
On Mon, 19 Aug 2019 at 15:25, Răzvan Crainea wrote:
> Hi, Solarmon!
>
> The parameter you should use is exactly the one you are using,
> rtpproxy_disable_tout[1]. That parameter says that after OpenSIPS
> detects the node as being down, it re-tries to send them requ
Hi,
Can somebody clarify when the rtpproxy status and health checks are done
and what configuration is required.
I am finding that the status/health of an rtprpoxy node is only
done/checked during opensips startup or rtpproxy module config reload. If
the rtprpoxy node goes down or comes back up,
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