| Summary: | ASTERISK-02163: DTMF to SIP channel sent too soon | ||
| Reporter: | flydoc (flydoc) | Labels: | |
| Date Opened: | 2004-07-31 17:55:42 | Date Closed: | 2011-06-07 14:10:10 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Core/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | nnnn,1,Answer nnnn,2,SendDTMF(1234567890) nnnn,3,Hangup using rfc2833 (easier to see problem) will send 1st digit after SIP TRYING message but before 200OK and ACK receiving device not ready at this stage ****** ADDITIONAL INFORMATION ****** adding Wait(1) before SendDTMF fixes problem I belive this may be a general RTP problem | ||
| Comments: | By: Mark Spencer (markster) 2004-07-31 21:54:53 Then don't send audio when the call's not up :) I don't think this constitutes a bug. By: Digium Subversion (svnbot) 2008-01-15 15:04:06.000-0600 Repository: asterisk Revision: 3552 U trunk/include/asterisk/module.h U trunk/rtp.c ------------------------------------------------------------------------ r3552 | markster | 2008-01-15 15:04:06 -0600 (Tue, 15 Jan 2008) | 2 lines Don't hard code the RTP payload type to 101 (bug ASTERISK-2163) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=3552 | ||