Hi,
I have an Asterisk 1.2.2 box (it was 1.2.1 just minutes ago) and
I?m trunking against a Lucent Softswitch (known as LSS) using SIP.
I have a Cisco 7960 as an * client; when I dial to PSTN the
Asterisk box sends the INVITE to the LSS but the problem is that the
LSS replies with the 100 Trying and then sends a 183 progress tone
(with SDP and RTP stream) and then sends a 180 Ringing; both messages
are forwarded to the 7960 so I end with two progress tones on my 7960
the local 180 Ringing generated by the phone and the C.O progress tone
sent by the LSS/MG.
I don?t know if is correct to send multiple 18X in reply to a SIP
INVITE but probably receiving a 183 message should cancel or supress
any subsequent 180 messages in the received endpoint. Is there anyway
to set this 180 supressing in Asterisk or sending 180 after 183 is not
allowed on RFC3261 and LSS is horribly broken ? Or maybe the 7960
should honour the 183 and suppress the 180 ?
TIA
--
Juanjo sin .sig :(