[asterisk-users] problems with natted phones

Abdenasser Ghomri ghomri.nasser at gmail.com
Thu Jul 8 14:09:15 CDT 2021


Have you tried to see if the router has any SIP ALG enabled this might
create such issues, Thanks.

Best Regards,


On Thu, Jul 8, 2021, 19:59 Marek GreŇ°ko <mgresko8 at gmail.com> wrote:

> Hello,
>
> I have an asterisk setup using pjsip. Everything used to work
> correctly until one remote site changed internet provider and thier
> router does not support sip protocol algorithms.
>
> It works for some time, but then suddenly audio stops working both
> directions. When this happens I see RTP responses going out to the
> local address of the natted phone, not to the natted address. The
> problem appears for the phones independently.
>
> The asterisk is connected to the internet with public static IP address.
>
> The pjsip config contains:
>
> [aor]
> type=aor
> qualify_frequency = 60
> max_contacts=1
> remove_existing = yes
>
> [endpoint]
> type = endpoint
> context = internal
> dtmf_mode = rfc4733
> disallow = all
> allow = alaw
> allow = ilbc
> allow = g729
> allow = gsm
> allow = g723
> direct_media = no
> allow_subscribe = yes
> subscribe_context = blf
> rewrite_contact = yes
> rtp_symmetric = yes
> force_rport = yes
>
>
> Am I missing something? Why the communication breaks suddenly?
>
> Thanks
>
> Marek
>
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