Summary:	ASTERISK-17676: audio dropped on attended transfer if first call uses g722
Reporter:	Joao Carvalho (foxfire)	Labels:
Date Opened:	2011-04-12 04:23:36	Date Closed:	2011-07-26 14:21:40
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/Transfers
Versions:	1.6.2.17	Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) log.txt
Description:	I consider this to be medium Severity. If g722 is not used on an attended transfer no problem everything works fine. But if g722 is used in the first call there is a problem. let us take for example user A,B and C. A calls B ( A uses g722 and B does not have g722 ) B starts the attended transfer and dials to C while it is ringing and C hasn't answered i get the following warning: [Apr 11 09:40:58] NOTICE[9507]: channel.c:3137 __ast_read: Dropping incompatible voice frame on Local/3001@enumtrans-1628;2 of format g722 since our native format has changed to 0x4 (ulaw) but no problems in audio just the anoying message constantly in the console. C answeres and no more warning messages all is Ok. Finaly B does a hangup to transfer the call and the real problem apears audio only in one direction and the following error message: [Apr 11 09:40:58] NOTICE[9507]: channel.c:3137 __ast_read: Dropping incompatible voice frame on Local/3001@enumtrans-1628;2 of format g722 since our native format has changed to 0x4 (ulaw) i have noticed that when using the Local channel there are some problems i have tried to avoid local channels in the dialplan , but in this case of an attended transfer i can't avoid it. By the way i used a Grandstream as A and Polycom as B and C. Please let me know which logs you require.
Comments:	By: Joao Carvalho (foxfire) 2011-04-12 04:26:33 i added a normal console log for now so you can get the general idea. By: Joao Carvalho (foxfire) 2011-04-13 03:28:54 UPS the second error is wrong in my description above.i did paste the wrong line. The error is like found in the log : Apr 11 09:41:12] WARNING[9529]: chan_sip.c:6290 sip_write: Asked to transmit frame type 64, while native formats is 0x1000 (g722)(4096) read/write = 0x40 (slin)(64)/0x1000 (g722)(4096)     sorry about the mixup By: Russell Bryant (russell) 2011-07-26 14:21:30.908-0500 Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions If this is still an issue, please open a new issue so it can be re-triaged appropriately. Thanks!