I have a problem with one way audio on Sip and I guess it may be a NAT
issue, in the example below 204 is rung by 208 (xlite external)
I dial perfectly but when I get to the answering of the Asterisk, I can
hear audio from the Asterisk but cannot get audio to the Asterisk, ie If
I ring the voice mail , Asterisk answers and then cannot hear my
password...
I have put the Ports Forward etc...5004-5080 & 10000-20000
Any ideas - even what to test next would be good...
-- Executing [...@macro-stdexten:13] Dial("SIP/208-00a10004", "SIP/204")
in new stack
-- Called 204
-- SIP/204-00a11584 is ringing
-- SIP/204-00a11584 answered SIP/208-00a10004
[Jul 7 16:15:08] WARNING[834]: chan_sip.c:1993 retrans_pkt: Maximum
retries exceeded on transmission
NjQ1NTA3Mzg3YzI5ZTNlYzI3MWU2NzE4YWE3ZTI2MDE. for seqno 2 (Critical
Response)
[Jul 7 16:15:08] WARNING[834]: chan_sip.c:2017 retrans_pkt: Hanging up
call NjQ1NTA3Mzg3YzI5ZTNlYzI3MWU2NzE4YWE3ZTI2MDE. - no reply to our
critical packet.
== Spawn extension (macro-stdexten, s, 13) exited non-zero on
'SIP/208-00a10004' in macro 'stdexten'
== Spawn extension (macro-stdexten, s, 13) exited non-zero on
'SIP/208-00a10004'
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