Is anyone out there using an "AudioCodes MP108 8-Port FXO Analog Gateway (SIP)" with asterisk to support both inbound and outbound calling? If so, I'm interested to hear how it works, and I'd love to see some example confs (both in sip.conf and on the MP108).

This product has been recommended to me by a SNOM/Asterisk-friendly distributor, but I would like a second opinion before purchasing. Incidentally, if this works well it could be an option for those of you who have recently asked about putting multiple analog FXO ports into an asterisk server.

Thanks,
--Ernest W. Lessenger
OACYS Technology

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