Summary:	ASTERISK-09506: Dial option S and reinvites
Reporter:	Gaspar Zoltan (gasparz)	Labels:
Date Opened:	2007-05-24 09:12:14	Date Closed:	2007-06-06 08:32:51
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Applications/app_dial
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) sip_log.txt
Description:	Hi, We are trying to setup asterisk to send reinvites and use the max_duration option for Dial (option S). All goes well but when the max_duration is passed after the dial we cannot play any sound: WARNING[27869]: file.c:587 ast_readaudio_callback: Failed to write frame. Looking into this issue we found out that when the call is hanguped by asterisk the RTP ip is not taken back to asterisk by another invite, so it makes sence failing to send RTP. Our testcase: We use agi for running the applications but I really think this is not affecting the test. The succesion of the applications run for this test is: Answer say_number(12) Dial(Sip/destination@provider||S(5)) say_number(34) hangup We captured the sip messages and the callflow is: Phone -> asterisk: invite Asteriks - >Phone: 100 Trying Asteriks - >Phone: 200 OK Phone -> Asterisk: ACK (now on the rtp we have the the say_number(12) Asterisk -> Provider: Invite -> SDP: RTP endpoint Asterisk ip Provider -> Asterisk: 100 Trying Provider -> Asterisk: 183 Session Provider -> Asterisk: 200 OK Asterisk -> Provider: Ack Asterisk -> Phone: Invite -> SDP: RTP endpoint Provider ip Asterisk -> Provider: Invite -> SDP: RTP endpoint Phone ip Provider -> Asterisk: 100 Trying Phone -> Asterisk: 200 OK Asterisk -> Phone: Ack Provider -> Asterisk: 200 OK Asterisk -> Provider: Ack (I assume right now the RTP goes between the phone and the provider on the (newly) negociated codec and have 5 seconds of call) Asterisk -> Provider: Invite -> SDP: RTP endpoint Phone ip Provider -> Asterisk: 100 Trying Provider -> Asterisk: 200 OK Asterisk -> Provider: Ack (I didn't understand compleatly why it sent an identical invite the asterisk but it doesn't hurt) Asterisk -> Provider: bye (the 5 seconds have passed) Provider -> Asterisk: 200 ok (I assume now we have the WARNING[27869]: file.c:587 ast_readaudio_callback: Failed to write frame.) Asterisk -> Phone : bye Phone -> Asterisk : Ok My opinion is that fixing this issue is sending another invite to the phone with SDP: RTP ip address to asterisk if a reinvite had been sent out. This way the playbacks and any other RTP after the dial can work. attacked are the actual sip messages.
Comments:	By: Joshua C. Colp (jcolp) 2007-06-06 08:32:50 Fixed in 1.2 as of revision 67649, 1.4 as of revision 67650, and trunk as of revision 67651. Thanks!