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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