I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.
Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.
The dialplan is real easy:
[from-teliax-sip]
exten => _j.,1,NoOp("From teliax sip with exten "${EXTEN}")
exten => _j.,n,Set(3digitexten=${EXTEN:12:3}
exten => _j.,n,NoOp("Callerid is " ${CALLERID(all)} )
exten => _j.,n,GoTo(from-outside,${3digitexten},1)
[from-outside]
exten => 123,1,NoOp()
exten => 123,n,Answer()
exten => 123,n,Dial(SIP/jnctn/1212xxxyyyy)
exten => 123,n,HangUp()
sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no ; tried nonat
sip show peer jnctn:
Insecure : invite
Force rport : Yes
.........
DirectMedia : No
sip show peer teliax:
Insecure : port,invite
Force rport : Yes
........
DirectMedia : No
And the cli doesn't show any problems:
NoOp("SIP/teliax-00000022", ""From teliax sip with exten
"<somename12lg>(123)"") in new stack
Set("SIP/teliax-00000022", "3digitexten=123") in new stack
NoOp("SIP/teliax-00000022", ""Callerid is " "") in new stack
Goto("SIP/teliax-00000022", "from-outside,123,1") in new stack
-- Goto (from-outside,123,1)
NoOp("SIP/teliax-00000022", "") in new stack
Answer("SIP/teliax-00000022", "") in new stack
Dial("SIP/teliax-00000022", "SIP/jnctn/1212aaabbbb") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaabbbb
-- SIP/jnctn-00000023 is making progress passing it to
SIP/teliax-00000022
-- SIP/jnctn-00000023 answered SIP/teliax-00000022
-- Locally bridging SIP/teliax-00000022 and SIP/jnctn-00000023
== Spawn extension (from-outside, 123, 3) exited non-zero on
'SIP/teliax-00000022'
The called party can hear the calling party, but not the reverse!
Any help really appreciated!
sean
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