We've had an issue since we went live nearly two years ago on Asterisk where people complain about not being able to talk while someone else is talking. I had assumed for a very long time this was because of the phones we went live with (Grandstream GXP-2000's) and for the longest time I believed this was a speakerphone problem only.
Last week during budgets, a request to buy new phones was put in to fix this problem. It was then that I finally researched and found that our phones do in fact support full-duplex on the speakerphone and handset. So, I'm looking for where I may have missed, perhaps an option on the Dial() command? Something in the phone config? We use SIP for phone to phone conversations, IAX for site to site conversations and Zaptel for PSTN lines. I've been told it happens regardless of protocol, so I assume it's one of the above options. Thanks for any help, Ken
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