Dinesh,
I suspect your problem is with the softphone implementation...
I have an Asterisk PBX setup with ISDN (chan_capi) and use Cisco 7960 phones
with Cisci SIP 7.5 firmware and get to watch the various SIP messages in/out
on the phone.
Depending on the phone numbers I dial (and the signalling back from the ISDN
exchange) I get 100 -> 183 -> 180 or 100 -> 180
In both cases the Cisco plays our ringing on receipt of the 180.
Occasionally calls which go from 100 -> 180 without going via the 183 result
in the Cisco ringing and combined rining genrated by the telephone exchange
which is weird but ok.
I have also encountered (rarely) ISDN number which, when dialled from 100 ->
183 -> Connected without a ringing phase - these call result in silence at
the Cisco phone followed by connected audio (from the far end) - which is to
be expected.
Mike
----- Original Message -----
From: "Dinesh Nair" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Sent: Tuesday, August 15, 2006 7:18 AM
Subject: [asterisk-users] Sending SIP 183 Session Progressing
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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