On 9/30/07, li zhang <[EMAIL PROTECTED]> wrote: > I find a issue that the audio sometimes may interrupt when talking and > can not listen anything since then unless you hangup and redial. I use > WireShark and find rtp stream still send/receive ok. The revision of used > sipXtapi is 10031 and audiocodec is gsm. > Anybody encounter this problem? I wish for your help!
This may be caused by bug, fixed in svn rev 10241: http://scm.sipfoundry.org/viewsvn/sipX?view=rev&rev=10241 You may safely backport this fix to rev you're using. -- Regards, Alexander Chemeris. SIPez LLC. SIP VoIP, IM and Presence Consulting http://www.SIPez.com tel: +1 (617) 273-4000 _______________________________________________ sipxtapi-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/
