I forgot to mention I had sent profiles several times before over a period of several months on all the sipX components in an attempt to fix the previous problems, to no avail. What essentially fixed the issue was clearing out and re-entering all of the intranet information in System>>Internet Calling and then resending profiles on everything.
Tony Graziano wrote: > > > On Sat, Oct 31, 2009 at 5:58 PM, Josh Patten <[email protected] > <mailto:[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]> > >> <mailto:[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]> > >> <mailto:[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] > <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/ > > 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. > > _______________________________________________ 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/
