On Thu, May 19, 2022 at 3:52 AM Dovid Bender <do...@telecurve.com> wrote:
> David,
>
> Are you getting any RTP from the PSTN for either leg? If not it could be
> that they assume you are behind NAT and want to see where the SRC of the
> RTP before they send it back. We had a few carriers that did this. The
> easiest way to get around it was to play a 0.5 second audio clip to the
> incoming leg. This will send RTP to the inbound carrier, causing them to
> send RTP back to you which would then hit the terminating carrier, which
> then sends you back RTP completing the loop. The dialplan looks
> something like this.
>
> same => n, Progress()
> same => n,
> Playback(/var/lib//asterisk_custom/sounds/xc,noanswer)
> same => n, Dial(SIP/+${EXTEN}@carrier,,)
>
I've also seen this happen due to networking equipment, specifically the
equipment wanting Asterisk to send packets before allowing packets in. If
both sides of the call are in this state, then you reach a stalemate and
media won't flow. Since rtp_keepalive is generated by Asterisk, it gets
sent, and media starts flowing.
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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