Hi Community,
I have a doubt, how would be the configuration of the Dial peer (pots) to connect to an analog trunk (Fxo) or an E1 link.
The type of configuration is a Local Gateway with an Existing Unified CM Environment. (Co resident PSTN-Local Gateway)
I see a voice class uri (100) is defined in the documentation
voice class uri 100 sip
host ipv4:192.168.80.13
Later, the outgoing dial peer (voip) is configured to the pstn, pointing to the carrier's ip (ITSP)
dial-peer voice 101 voip
description Outgoing dial-peer to IP PSTN
destination-pattern BAD.BAD
session protocol sipv2
session target ipv4:192.168.80.13
voice-class codec 99
dtmf-relay rtp-nte
voice-class sip tenant 100
no vad
What would that configuration be like, but towards an analog trunk (Fxo) or an E1 link?
In case there is no sip trunk service from the carrier
