The codecs are:
SIP Phone:
choice 1: PCMU
choice 2: PCMA
choice 3: G723
choice 4: G729
choice 5: G726-32
choice 6: G728
Asterisk:
in sip.conf
1: ulaw
2: alaw
in oh323.conf
1: G711U
Gateway:
preference 1: G711U
preference 2:
.
.
.
preference 8: G711A
That's good? Can you see where's the
Mireia Munoz de jesus wrote:
The codecs are:
SIP Phone:
choice 1: PCMU
choice 2: PCMA
choice 3: G723
choice 4: G729
choice 5: G726-32
choice 6: G728
Asterisk:
in sip.conf
1: ulaw
2: alaw
in oh323.conf
1: G711U
Gateway:
preference 1: G711U
preference 2:
.
.
.
preference 8: G711A
Try with
The call end reason EndedByQ931Cause is used by the OpenH323 stack when it
doesn't know the real cause.
Try to see if the codecs in the gateway are compatible with the codecs in
asterisk.
What are the codecs you are using in SIP Phones, in Asterisk and in the
gateway?
Regards,
Vinicius