i'm not sure if this is a -users or a -dev question, but am sending it here
anyways. discussion could move to -dev if chan_sip.c code needs to be
amended/explained.
first up, all this on asterisk 1.2.10 on freebsd 6.1.
here's the beef:
from a particular sip softphone we're playing with, we notice that calls to
another SIP phone (same LAN) result in the /lack/ of a ringing tone on the
softphone. however, calls from the same softphone to a PSTN/Mobile number
(through a TE405P) result in proper behaviour on the softphone with a
ringing tone.
an ethereal trace of both types of calls results in only one difference.
for calls to the PSTN/Mobile thru libpri/chan_zap, asterisk returns a SIP
183 Session Progress[1] packet in between the 100 Trying and 180 Ringing,
while for calls from the softphone to another SIP phone it's 100 Trying
followed immediately by 180 Ringing.
so my question is, is the softphone behaving correctly in not playing a
ringing tone to the user without the 183 packet inspite of the 180 Ringing
packet being received ? alternatively, since we aren't able to change the
softphone, will i break anything big if i force asterisk to send the 183
packet immediately after sending the 100 Trying packet in sip_indicate() ?
alternatively, in reading the RFCs, i came across RFC3398 which speficies
mappings between ISDN Cause Codes and SIP responses. has this mapping been
implemented in asterisk at the moment, either in 1.2 or the upcoming 1.4 ?
[1] the 183 Session Progress packet is triggered by the receipt of a PRI
PROGRESS indicator from libpri, which gets translated to a
AST_CONTROL_PROGRESS and thence a 183 Session Progress to SIP.
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED] (0 0) http://www.openmalaysiablog.com/
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