I commented about this in the jira. sipx is not sending the packet back
to itself. It looks like at first glance but you need to understand the
the bridge proxies rtp. Thus, the packets you see that look like they
are looping back are packets from the ITSP towards sipXbridge' media
proxy then towards the IVR. What's I see is that after the IVR plays
the prompt for the voice mail, it stops sending audio. There is a
parameter in the managed gateway configuration that allows you to
specify NAT keep alive for RTP. Try using the "replay" option if that
solves the issue. I have also sent the message below in both freeswitch
irc channel and their mailing list. Maybe someone here knows the answer
here.
-----
Sent to FS user list:
I have recently received a packet capture from the sipx community that
seems to indicate that when FreeSwitch is recording a voice mail message
via our custom IVR, that it does not send either comfort noise or
silence packets back to the caller. This results to the call getting
dropped by the ITSP after 30 seconds because of RTP time out. Is there a
configurable parameter to make the IVR send some voice activity back?
----
On 11/12/2011 08:14 PM, Tony Graziano wrote:
packet 6752 is comfort noise, sent from sipx back to itself? i would
think the comfort noise would need to be sent to the trunk provider.
if the rtp keepalive loops back to sipx, its not going to keep it alive.
On Fri, Nov 11, 2011 at 6:14 PM, Gerald Drouillard
<[email protected] <mailto:[email protected]>> wrote:
On 11/11/2011 5:36 PM, Tony Graziano wrote:
> I also decided to test it with other media services (conferencing),
> and it only seems to be with voicemail. So I suspect sipx needs a
> freeswitch tweak to get around this. Since other media services (AA,
> conferencing, faxetc.) have rtp going the other way, it makes me
> suspect its a voicemail setting (freeswitch parameter), and I recall
> there was a "proxy media" setting for freeswitch invoked to handle
> fax, so I suspect the media settings in freeswitch need to be
revisited.
Appia has no problem leaving an a voicemail over 1 min to the
exact same
mailbox. I just find it strange that voip.ms <http://voip.ms>
would hang up the call
with media streaming in the one direction. You would think that it
would at least ask for a "are you still there?" response before
hanging up.
A cap file for anyone interested can be found at:
http://www.drouillard.biz/tmp/test.cap
--
Regards
--------------------------------------
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
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--
======================
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Telephone: 434.984.8430
sip: [email protected]
<mailto:[email protected]>
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Email: [email protected] <mailto:[email protected]>
LAN/Telephony/Security and Control Systems Helpdesk:
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