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

[Steve Totaro]

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)