Were holding ourselves back from moving to PJSIP as we dont appear to have figured out how to force codec preference in a dial plan. The PJSIP Advanced Codec Negotiation document (https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiati..
Author : David Herselman
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We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial inv..