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

[Eric Wieling]

You will find not transcoding much less useful that one might imagine. 


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of d tbsky
Sent: Thursday, September 25, 2014 2:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Change codec when dial from SIP to DAHDI

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!!

Regards,
tbskyd

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