Paul Edgar
2009-Jul-08 01:55 UTC
[asterisk-users] One Way Audio from External Sip Soft & Hard Phone
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' -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090708/e8bb0b84/attachment.htm
Steve Totaro
2009-Jul-08 02:34 UTC
[asterisk-users] One Way Audio from External Sip Soft & Hard Phone
On Tue, Jul 7, 2009 at 9:55 PM, Paul Edgar<paul at 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 [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' > >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)