Try setting the channel var record_waste_resources=true in the dp before calling VM or globally in vars.xml This will cause useless silent RTP to be fired at the caller while they are recording. Without this it logically stops sending while its recording the message since you don't need to hear anything while you are recording your message. Some devices panic with no media and hangup thus this workaround.... On Fri, Apr 19, 2013 at 11:35 AM, Michael Collins <msc@freeswitch.org>wrote:
This is interesting. I suspect she has given you only the facts as she sees them and not necessarily all of the facts. I believe you'll need pcaps of the calls in question to know for sure.
As far as FreeSWITCH being able to tell the difference between a human and a voicemail/answering machine... not without a special module. :)
Get those pcaps and FS logs and see if there are any clues. If you can compare a working call against a non-working call then that might help you narrow down where to look.
-MC
On Fri, Apr 19, 2013 at 7:55 AM, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hi All,****
** **
I have a report by a customer at a site that we had considerable trouble with NAT issues. I thought everything was now stable, but it sounds like it is not.****
** **
She reports that she is only telling me now, but for some time when they call someone and get no answer and get directed to voicemail (the latest was a Sprint number) the audio from the remote answering machine cuts off before then end of the message. She did point out the call timer on her phone continued normally. She swears this does not happen when talking to a live person!****
** **
First, I don’t think FS can even tell if the b-leg was answered by voice mail differently from a person. Please correct me if that is incorrect.* ***
** **
So, I am proceeding on the assumption that calls that are answered take less than 60 seconds! 45 seconds of the reported call were waiting for Sprint to give up on the cell phone and route to voicemail.****
** **
The log just shows the call being answered … then Normal Clearing.****
** **
I tend to always suspect NAT or crappy Polycom 335s for this customer as that has been all my issues with them so far. Is there a NAT issues relating to max call length, etc? No other users (all cisco phones) on the same FS at other sites report this issue.****
** **
Thanks in advance.****
Sean****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
