On Saturday 10 July 2021 at 22:57:09, Eric Wieling wrote:
> > On Sat, Jul 10, 2021 at 2:39 PM Alexander Perkins wrote:
> >
> > Hi All. We have a provider that requires us to SOURCE the SIP
> > connection on TCP 5061. I honestly have no clue how to force
> > Asterisk to always SOURCE
Kamailio is useful when you want to do weird, non-standard, or unusual
stuff with SIP. You could send your outgoing connections to Kamailio,
which could then send the connection out with the required source port.
Have you considered using a not stupid provider?
On 7/10/21 3:44 PM, Joshua C.
On Sat, Jul 10, 2021 at 2:39 PM Alexander Perkins <
alexanderhenryperk...@gmail.com> wrote:
> Hi All. We have a provider that requires us to SOURCE the SIP connection
> on TCP 5061. I honestly have no clue how to force Asterisk to always
> SOURCE the SIP connection on a certain port.
>
> Can
I don’t think I’ve seen that requirement before, so someone else may have to
answer if there is a PJSIP specific setting
However, if not then it may be simple to achieve the same result by using your
firewall NAT rules.
From: asterisk-users
Hi All. We have a provider that requires us to SOURCE the SIP connection
on TCP 5061. I honestly have no clue how to force Asterisk to always
SOURCE the SIP connection on a certain port.
Can anybody point me in the right direction? I am using PJSIP.
Thank you,
Alex
--
Hello,
I just disabled. Currently it is working. I shloud give it some time
to confirm the problem has gone. Maybe one month would be enough to
confirm.
Thanks
Marek
2021-07-09 20:11 GMT+02:00, Abdenasser Ghomri :
> Yes just disable the SIP ALG and see if it helps, Thanks.
>
> Best Regards,
>
