On Sat, Oct 31, 2009 at 5:58 PM, Josh Patten <[email protected]> wrote:
> Well it seems I found out why this was happening, and it was related to > sipXbridge and the media relay. > > Even though I had my domains and subnets defined in System>>Internet > Calling, sipX was sending all my sip and media session to my sipX > installations' external address (1-to-1 NAT) which then sent them to the > necessary devices. The reason this was happening is because for some > reason neither my intranet domains or subnets settings had actually > taken effect since the day I installed the system. Here is what I did to > fix the system: > > Go to System>>Servers>>"server.domain.tld">>Confgure. From there I > disabled SIP Trunking and after that I resent the server profile and > restarted the server. At that point I was still having media path issues > with an Asterisk server that I host some custom applications on. I > changed the RTP port values in the NAT area of the server and then went > and redid all of my intranet settings in System>>Internet Calling, > resent server profiles, re-enabled SIP trunking (after installing > patch17), resent server profiles, and restarted the server completely > (warm reboot). After doing all of that I was able to make a 1.5 hour > phone call over a VPN connection with no audio issues or hangup issues. > > > Josh Patten wrote: > > OK, here is my snapshot and tcpdump (links, no file attachments): > > > > > http://www.filefactory.com/file/a07ef89/n/sipx-snapshot-it_ippbx_co_brazos_tx_us_tar_gz > > > > http://www.filefactory.com/file/a07egb8/n/sipx-tcpdump_cap_bz2 > > > > Seems now the calls are being one-wayed anywhere from 25-35 minutes, > > time varies on each call. > > > > Happy hunting. Thanks for your time. > > > > M. Ranganathan wrote: > >> > >> > >> On Tue, Oct 27, 2009 at 11:49 PM, Josh Patten <[email protected] > >> <mailto:[email protected]>> wrote: > >> > >> It's is the weirdest thing; Using sipXbridge between sipX and my > >> Adtran > >> TA 908e, which is on the same subnet as my sipX installation, > >> randomly > >> causes one-way audio. I am using sipXbridge patch17. The original > >> 4.0.2 > >> sipXbridge would only occasionally have a one way audio problem, but > >> when I implemented patch14 I had a lot more issues with one way > >> audio > >> and dropped calls, and patch17 made the one way audio consistent for > >> each call albeit at different times throughout the call. > >> Sometimes the > >> call would go for 10 minutes before going one way audio, sometime > >> 3 minutes. > >> > >> Here is something I did that made the audio last for 40 minutes > >> on one > >> call: under system>>servers>>"name.of.server">>Services>>Media > >> Relay I > >> changed "Media Relay Temperament" from conservative to aggressive. > >> > >> Can someone give me a clue what I should do next? > >> > >> > >> > >> > http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration#Problem_reporting > >> > >> > >> > >> > >> You can mail me a snapshot. I will look as time permits. > >> > >> Ranga > >> > >> > >> _______________________________________________ > >> sipx-users mailing list [email protected] > >> <mailto:[email protected]> > >> List Archive: http://list.sipfoundry.org/archive/sipx-users > >> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > >> sipXecs IP PBX -- http://www.sipfoundry.org/ > >> > >> > >> > >> > >> -- > >> M. Ranganathan > >> > > > > _______________________________________________ > sipx-users mailing list [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > Any changes (like installing a patch), changing a gateway (ITSP profile) or SBC, has a function to "send profiles", The reason this is there is because it is "plannable" by the admin as to when to do this, because then some services will prompted to restart, and unloike a separate hardware gateway, will break any call in progress to a remote worker or connecting to an ITSP via the built in SBC. You have to decide when to do this, of course. Also, I think the instructions for patch17 should include what services to restart (sipXconfig (because of a running java process, and sipxbridge). I also noticed changing a dialplan with a remote user connected caused the registration to drop when testing a Bria Pro configuration. I think if I had given it a few seconds it might have reconnected, but I was trying to get through some tests so I just restarted the PC app. I think that's a functionality within the Bria and not sipx specifically. It's just the way it is.
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