Hi all,
I'm running into some issues with E164 conversions and was wondering if there
was a decent standardized method for conversions. Conversions using $rU seem to
be working great, but $fU is behaving badly. This is what I have so far:
xlog("$ci - caller id was $avp(src=>callerid)");
Hi Duncan,
This scenario is quite new for me, not sure I got it right.. but why have
you decided not to proxying requests to asterisks?
By leveraging Path and Record-route headers Asterisk will know how to route
the response back as well as new requests.
And the proxy will know how to handle them.
Hi there
I am a new user of Kamailio and we are trying to use it to be as a front
end for our asterisk pbx. We are running on Ubuntu 18.04 and Kamailio 5.3.8
with Siremis
Rather than proxying the request through to asterisk we are trying to use
uacreg to send a login to asterisk. Asterisk will th
Correct--that it when I see the "qm_free(): BUG: freeing already freed
pointer" error. -A
On Tuesday, July 27, 2021 9:05:45 AM CDT Daniel-Constantin Mierla wrote:
> Hello,
>
> to be clear for me: the error messages pop up only when outbound module
> is used and only for branches that are routed
Hello Abdirahman,
There’s no support for “defines” for the xhttp_pi file.
Regards,
Ovidiu Sas
On Tue, Jul 27, 2021 at 11:34 Abdirahman Osman
wrote:
> Hello everyone,
>
> In the xhttp_pi file of pi_framework.xml,is there a way I can pass the
> password to the xhttp_pi module as a pre-processing
Hello everyone,
In the xhttp_pi file of pi_framework.xml,is there a way I can pass the
password to the xhttp_pi module as a pre-processing define?
postgres://kamailiorw:kamailiorw@localhost
/kamailiorw
Thank you!
Abdirahman
__
Kamail
Hello,
to be clear for me: the error messages pop up only when outbound module
is used and only for branches that are routed using outbound-specific
connections?
Cheers,
Daniel
On 25.07.21 04:55, Anthony Joseph Messina wrote:
> I've come a bit further to find how to reproduce...
>
> I have a sin
Hello,
are you using topos module?
If yes, can you try without it and see if there is any difference?
Also, the sip trace you pasted in the previous message does not have
both incoming and outgoing sip messages -- having them is useful for
troubleshooting.
Cheers,
Daniel
On 23.07.21 22:16, Joã
Hi,
Some comments on Yate.
Yate works well as a B2BUA. It is very performant and stable. It can bypass
media dynamically.
The main beauty of Yate is how it processes calls internally. Yate represents
calls internally as messages with parameters (caller, callee, codecs …) . While
the call p
Hello,
just for the records:
You can also bypass media with FreeSWITCH if you want. The overall handling
of many parallel calls is still not so high like in SEMS.
https://freeswitch.org/confluence/display/FREESWITCH/Proxy+Media
This is also possible on a per Call base.
And you can use different
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