Please can someone confirm if alias_db_lookup() and alias_db_find() are
blocking functions.
If they are, can they be used in the async function?
I'm using v2.4.6 and 2.4.7 of OpenSIPS.
Thanks.
John Quick
Smartvox Limited
Web: www.smartvox.co.uk
___
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
Hi
I have complied RTP Proxy and its working also I have added rtpproxy module
in opensips.cfg file but it is not working.
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Smells like a OS/kernel bug to me. There is little application can do in
that regard, UDP fragmentation/reassembly happens at much lower layers of
the OSI stack.
However, as a workaround as long as SIP goes you can try to reduce your SIP
signalling packet size by using compact version of SIP
Hey Michael,
Maybe this {ip.isprivate} is a better options:
https://www.opensips.org/Documentation/Script-Tran-3-1#toc80
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
On 5/18/20 8:29 PM, Saint Michael wrote:
I need to identify all
Hi,
We have an issue on our home proxy (opensips 2.4.6), when it receives 200
OK (over UDP) from our Freeswitch and the package size is higher than the
MTU size , we sometimes get fragmentation of the UDP packets, but only the
first part of the fragmented package is sent to our edge proxy.
My script has
if (is_method("INVITE") && !has_totag() && check_source_address(0)) {
xlog("[ROUTE]Incoming call to MS: RURI=$ruri, SI=$si, M=$rm\n");
trace("tid");
create_dialog();
do_routing(1);
strip(1);
prefix("+44");
Whenever I have tried to run OpenSIPS on the same host server as either
Asterisk or FreeSwitch (using different ports), I have always hit a snag
with routing of sequential loose-routed requests. Possibly this only happens
when using double Record-Route headers which I need for protocol conversion
I need to identify all private IPs vs public IPs
Right now I am doing
$rd =~ "192.168" || $rd =~ "10." || $rd =~ "172.16."
but in regex there is a faster way, chaining several ORs
like
$rd =~ "192.168|10.|172.16."
what is the correct way to do this in opensips?
I thought I had just done some bad config. I have almost exactly the same
thing.
I have Freeswitch And Opensips on Azure VM’s
Freeswitch = 10.0.0.4 + External IP
Opensips = 10.0.0.5 + External IP
Both are configured to use the external addresses
Freeswitch will start the conversation on its
There are several calling scenarios – typical Class V – where multiple
SIP dialogs may be involved. And to make it work, you need, /from one
dialog, to access the data that belongs to another dialog/.
https://blog.opensips.org/2020/05/18/cross-dialog-data-accessing/
Enjoy,
--
Bogdan-Andrei
Hi Mark,
First of all, all the upgrades are incremental, from one version to next
one, so you should do 2.4 - > 3.0 and 3.0 -> 3.1
Secondly, we haven;t yet prepared the migration docs/scripts for 3.0 ->
3.1, they will be available upon beta release.
Regards,
Bogdan-Andrei Iancu
OpenSIPS
Yes, I realise I might be getting a little ahead of myself :)
I think my question really should be 'can I upgrade the DB direct from 2.4
to 3.1 format?'
Mark.
On Mon, 18 May 2020 at 15:48, Giovanni Maruzzelli wrote:
> On Mon, May 18, 2020 at 4:19 PM Mark Farmer wrote:
>
>>
>> Just wondering
Hi Asteriskman,
As per doc [1], the module will provide the timestamp of the 200 OK
(call answering) . If doing "cdrs", you will also get the call duration,
already computed.
For anything extra, you should use extra accounting data [2].
The accounting engine does not natively support noSQL
Hi John,
Yes, these functions are performing DB queries in runtime and these
queries are potentially blocking.
The db_aliases module does not offer async support, but the queries it
is doing are trivial and you can do them via avp_db_query() + async()
support.
Regards,
Bogdan-Andrei
Hi Bogdan-Andrei,
Yes, I know, but I meant similar to $T_fr_timeout and
$T_fr_inv_timeout, which can be set in the script during message
processing.
Sorry if my questions wasn't clear enough.
Regards,
Grant
On Mon, May 18, 2020 at 1:49 PM Bogdan-Andrei Iancu wrote:
>
> Hi Grant,
>
> There are
Hi everyone
Just wondering if there's a 'supported' upgrade process to jump from 2.4 to
3.1?
Or will I need to do it in 2 steps (2.4 to 3.0 & 3.0 to 3.1)?
Many thanks
Mark.
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Hi Yuriy,
As you are using Kamailio and FreeSWITCH, it is more appropriate to post
the question to their mailing lists.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
On 5/14/20 6:09 PM, Yuriy Nasida wrote:
I use kamilio but I think it
Ah, gotcha no, there is nothing per transaction, these are global
options.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
On 5/18/20 2:51 PM, Grant Bagdasarian wrote:
Hi Bogdan-Andrei,
Yes, I know, but I meant similar to $T_fr_timeout
Hi,
Could you share further details as whats not working?
check the following:
1 - OpenSIPS is able to connect tot he rtpproxy socket
2 - OpenSIPS is calling the right rtpproxy function from the script
3 - the RTPproxy function have correct parameters
4 - Check the SDP for incoming leg and the
On Mon, May 18, 2020 at 4:19 PM Mark Farmer wrote:
>
> Just wondering if there's a 'supported' upgrade process to jump from 2.4
> to 3.1?
> Or will I need to do it in 2 steps (2.4 to 3.0 & 3.0 to 3.1)?
>
>
Ahem...
3.1 is not even in beta, I believe...
-giovanni
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