[asterisk-bugs] [Asterisk 0014249]: One way voice after attended transfer

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Jan 15 11:16:25 CST 2009


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14249 
====================================================================== 
Reported By:                RadicAlish
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   14249
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.21.2 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-01-15 04:49 CST
Last Modified:              2009-01-15 11:16 CST
====================================================================== 
Summary:                    One way voice after attended transfer
Description: 
We have problem with one way voice in next scenario:
101 calls to 102, 102 takes second line and calls to IVR (number 1234)
that have Answer() application and then call directed to 103
102 makes attended transfer while 103 is ringing, so now we have call
between 101 (connected) and 103 (ringing)
when 103 answers we get in CLI bulk of messages:
[15 12:33:10] WARNING[3681] chan_sip.c: Asked to transmit frame type 64,
while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x
8 (alaw)(8)
101 can hear 103, but 103 can't
only if 101 press any digit (sends DTMF) or press HOLD and UNHOLD, then we
have two way voice.

101, 102 and 103 are SIP devices
I attached full log with SIP debug
====================================================================== 

---------------------------------------------------------------------- 
 (0097931) RadicAlish (reporter) - 2009-01-15 11:16
 http://bugs.digium.com/view.php?id=14249#c97931 
---------------------------------------------------------------------- 
there is similar issue in another scenario:
101 calls to IVR (number 1234) that have Answer() application and then
call directed to 103, 103 doesn't want to answer and divert call (SIP 302
Moved Temporarily) to 102,
101 stops to hear ringing tones and we get in CLI bulk of messages:
WARNING[5475] chan_sip.c: Asked to transmit frame type 64, while native
formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
when 102 answers audio come back to 101
I attached another log full.issue_14249.2 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-15 11:16 RadicAlish     Note Added: 0097931                          
======================================================================




More information about the asterisk-bugs mailing list