[asterisk-users] One Way Audio from External Sip Soft & Hard Phone

[Paul Edgar]

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 [s at 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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