Asterisk has never had a complete enough SIP stack to be used as an unmanaged gateway. Use sipXbridge or a separate SBC. Or try FreeSWITCH.
On Fri, Aug 12, 2011 at 8:11 AM, Douglas Hubler <[email protected]> wrote: > On Fri, Aug 12, 2011 at 8:01 AM, cyril constantin < > [email protected]> wrote: > >> Call from sip peers on Asterisk to PSTN Avaya works without issue. >> Call from Sipxecs sip peers through Asterisk then using H323 connection to >> Avaya extension works >> Call from Sipxecs sip peers through Asterisk then using H323 connection to >> Avaya PSTN doesn't works, I can ring my mobile phone but when I answer the >> call I don't have the voice in both way. >> >> Codec used is G.711 Alaw. >> > > - check that's the final codec selected in all paths by analyzing the SDP > in SIP. check working and not working paths to ensure they are the same > - check specified RTP ip addresses and port numbers in the SDP of the SIP > messages. > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- Josh Patten eZuce Solutions Architect O.978-296-1005 X2050 M.979-574-5699
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