[asterisk-users] Change codec when dial from SIP to DAHDI

d tbsky tbskyd at gmail.com
Thu Sep 25 13:56:49 CDT 2014

2014-09-25 20:46 GMT+08:00 Matthew Jordan <mjordan at digium.com>:
>>     https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
> That article is in the development section of the wiki. While that
> doesn't mean any of the information there is necessarily wrong, its
> purpose was to coordinate development efforts, not to define behavior
> for end-users.
> In this particular case, portions of that page only affect chan_pjsip:

   thanks a lot for the hint! you really save my day!
   I was thinking about studying freeswitch, since people said
freeswitch can do that without transcode. now i will spent my time to
study chan_pjsip, and hope it can fix the problem. i really want to
stay with asterisk :)

   thanks again for your kindly help!!


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