Hi,
If you will send call without answering on asterisk and have directrtpsetup=yes
in sip.conf codec negociation will always be between UAs so any matched codec
will work fine. If you are answering call on asterisk then dialing it out to
next UA then you need to add canreinvite=yes for both UAs.
Regards,
Faisal
P peers calling each other:
A (g722, alaw) calls B (alaw,ulaw) via asterisk.
My setup:
allow=g722,alaw
preferred_codec_only=no
Note that when B calls A, codec alaw is used on both ends, fine, but it does not
seem to work the reverse way (A is using g722, B is using alaw, asterisk is
doing transcoding).
Is it possible?
Thanks,
Ondrej
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