-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Thursday 18 September 2003 19:04, Hielke Christian Braun wrote: > try to change [siptestphone] to [atrg613test] in sip.conf. Maybe > that helps.
It didn't. And now something else is weird. Asterisk fails sending audio to my SIP phone. Found this in my logs: Sep 19 11:08:52 WARNING[950291]: File channel.c, Line 1819 (ast_channel_make_compatible): No path to translate from SIP/sc.sc.sc.sc-de54( 4) to H323/ip$hc.hc.hc.hc:1244/14060(8) Sep 19 11:08:58 WARNING[147466]: File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call [hex]@ as.as.as.as for seqno 102 (Request) Sep 19 11:09:04 WARNING[147466]: File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call [hex]@ as.as.as.as for seqno 102 (Request) What on earth is this? Codec? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/as2r2TEAILET3McRAtIaAJ9Hpa3k/a7giiB62pwn7qw17jck/ACeJLdH fzoRqSVrEMfgAfzE5BOogoU= =N4hn -----END PGP SIGNATURE----- _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users