Hi, I have two asterisks that are connected to each other via a back-to-back E1 link using a pair of sangoma cards.
With the following scenario: SIP-PHONE <-> Asterisk <-> E1 <-> Asterisk <-> SIP-PHONE, the sound quality degrades significantly. I can't understand why as the amound of packet lost should be very minimum. Does anyone know why? Does it have anything to do with what codec to use? Thanks, Mark
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