I have an Audiocodes MP-114, in sip.conf I have two entire for PSTN line:
[pstn-5665] ; incoming/outgoing calls on FXO port 5665
type=friend
secret=xxxx
insecure=invite
username=fax-5665
host=dynamic
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
nat=no
context=incoming
callgroup=1
pickupgroup=1
[pstn-1270] ; incoming/outgoing calls on FXO port 1270
type=friend
secret=xxx
username=voice-1270
host=dynamic
insecure=invite
disallow=all
allow=ulaw
allow=alaw
nat=no
context=incoming
callgroup=1
pickupgroup=1
When call comes in on pstn-1270 asterisk shows in log "pstn-5665" eg:
-- Executing [11 at incoming:1] GotoIfTime("SIP/fax-5665-00753a80",
....
it should be "voice-1270"; when I process the calls using Linksys 3201
the entry is correct:
-- Executing [11 at incoming:1] GotoIfTime("SIP/fax-5665-00753a80",
Is Audiocodes sending the call to a wrong incoming line or asterisk is answering
with the wrong entry?
I cannot find anything in audiocdes logs that calls are going into wrong
incoming line.
--
Joseph