Summary:	ASTERISK-01854: Asterisk ringback tones on SIP channels
Reporter:	Anton Fedorov (datacompboy)	Labels:
Date Opened:	2004-06-19 23:11:18	Date Closed:	2008-01-15 15:00:54.000-0600
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) use180.patch.bz2
Description:	SIP protocol while ringing support reply  180 Ringing that cause phone to play ringback tone. But if need to indicate ringing tone for country that you call, not default -- you can't use that reply, need to ring via asterisk possibilities.  But for some devices / pstn termination services need to send 180 Ringing...  So I have added new configuration option into sip.conf -- "use180" (boolean) to general configuration, and to configuration for each user. use180 = yes  cause to indicate ringing via "180 Ringing" use180 = no  cause to indicate via "100 Ringing" and ringtone by asterisk Attached patch for latest CVS version.
Comments:	By: Anton Fedorov (datacompboy) 2004-06-20 00:25:25 OOoops, sorry, was posted into incorrect category. Should be SIP. By: Brian West (bkw918) 2004-06-20 10:46:37 Ok what does the RFC say because it all works correctly here.  Something in your Loop isn't doign it correctly.    Please attach sip debug so we can fix the problem properly. By: Mark Spencer (markster) 2004-06-21 14:06:50 His goal is to have Asterisk supply in-band ringing rather than just OOB ringing.  The right answer would actually be to supply 180 Ringing *and* return -1 thus causing the inband to be generated. By: Anton Fedorov (datacompboy) 2004-06-21 14:20:45 yes, RFC says to send "180". but that have mean only if remote is something that need to know that party is ringing. When remote is some man, who want to hear correct ringtone, better to use asterisks' indications -- but if asterisk send 180 AND generate indication -- then you he will hear "Beeeep. Beep-beep.", except of "Beep-beep" generated by asterisk. By: Mark Spencer (markster) 2004-06-22 00:05:55 I've added a "progress" application.  Try calling "Progress" before "Dial" and tell me if that does what you want. By: Anton Fedorov (datacompboy) 2004-06-22 02:58:25 Yes. Except of PSTN->SIP calls now hear no ringing tone, because to pstngates need to send 180 Ringing -- no voice passed before call begin. This why I have added use180 to config -- some peers need to be configured one way, other -- other way. By: Mark Spencer (markster) 2004-06-26 17:31:08 Okay I added it as "progressinband=yes" or "progressinband=no" which can be set per peer, user, or globally. By: Digium Subversion (svnbot) 2008-01-15 15:00:54.000-0600 Repository: asterisk Revision: 3324 U   trunk/channels/chan_sip.c U   trunk/configs/sip.conf.sample ------------------------------------------------------------------------ r3324 | markster | 2008-01-15 15:00:53 -0600 (Tue, 15 Jan 2008) | 2 lines Add option for in-band progress (bug ASTERISK-1854) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=3324