Hi Andrew Thanks for your response.
> Yes, the option is progressinband in sip.conf As far as I can tell, the progressinband setting does not prevent the 183 from being sent altogether. For example, with progressinband=never, early audio from UA1 to UA2 after a 180 ringing will still trigger a 183 from UA2 to UA3. > Why do you care about the 183 or not? The 183 message exercises a bug in the software used by a certain ITSP, whereby ring-back continues to be heard by a PSTN caller after the call is established with 200 OK. Disabling the 183 would be an effective temporary workaround. > Also it wouldnt hurt to read the SIP RFC's to have a better understanding of > what is going on: Thanks for the links, I have read these documents. Is there something I appear not to understand? Regards, Richard -- Richard Brady T: +44 (0)7771 623 348 E: [EMAIL PROTECTED] _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
