Progressinband=yes under [general] in sip.conf
NEXT!!!
- Joshua Colp.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dave DeChellis
Sent: Tuesday, December 28, 2004 8:57 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk / 183 message
Hello,
My company is doing some * testing with our Class 5 softswitch and had
some questions regarding ringback being provided to our PSTN users (off
--> on net calling)
Currently with MGCP subscribers, we know the PSTN ringing is provided by
a digital PBX for example, However, it looks like with SIP, our
softswitch is relying on MGCP signaling on our PSTN gateways to provide
ringback to the PSTN network - this makes complete sense (gateway to
PSTN gets a S: rg).
The problem is that some of our older gateways don't support MGCP event
signaling, so I was trying to get Asterisk to send all SIP calls to our
switch with a "183" message so maybe our switch will cut the call
through two-way immediately. Currently Asterisk is sending 180/ringing
messages to our switch.
Is there a way to have Asterisk send 183 messages to my peer defined in
sip.conf ?
Thanks,
Dave
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users