On Tue, Jul 7, 2009 at 9:55 PM, Paul Edgar<p...@tabs.co.nz> wrote: > 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' > >
Where is the NAT or is it on both sides? Answer that and turn on SIP debugging and post the output and I am sure someone can help you. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users