Re: [sipx-users] after update Openfire server won't start

Was able to install openfire.jar and it looks like it's running with no errors. I think I'll hold off on downgrading freeswitch and see how it works. So far, it's been working with ver 1.0.6 since the upgrade. Ly Tran From: Douglas Hubler Sent: Mon 8/2/2010 11:52 PM To: Tran, Ly V.; Gerald

Re: [sipx-users] after update Openfire server won't start

On Mon, Aug 2, 2010 at 11:10 AM, Douglas Hubler wrote: > Freeswitch 1.0.6 must have made it into 4.2.1 somehow for a time period. > There are no known issues, but it hasn't been tested so I'll figure out the > downgrade instructions later this evening, assuming I can afford the luxury > wifi in

Re: [sipx-users] after update Openfire server won't start

On Mon, Aug 2, 2010 at 3:16 PM, Tran, Ly V. wrote: > [r...@sipx ~]# md5sum /opt/openfire/lib/openfire.jar > md5sum: /opt/openfire/lib/openfire.jar: No such file or directory > > I am definitely missing the openfire.jar file.  Unfortunately, I can't try > the forced install of it until after closi

Re: [sipx-users] New to sipx - call coverage issues

From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of R. Benjamin Kessler [ben.kess...@zenetra.com] I didn’t get any responses to the below; I was hoping that somebody could point me in the right direction. __

Re: [sipx-users] New to sipx - call coverage issues

I don't think I understand. Are you setting the divert on the phone? What you should be doing is setting the Call Forward for a Polycom Phone in the sipxconfig UI. It will also make a difference how long the call forward is set for if it is using an analog gateway. Can oyu better explain how you

Re: [sipx-users] after update Openfire server won't start

[r...@sipx ~]# md5sum /opt/openfire/lib/openfire.jar md5sum: /opt/openfire/lib/openfire.jar: No such file or directory I am definitely missing the openfire.jar file. Unfortunately, I can't try the forced install of it until after closing. Or is there a way of doing it without having to stop/res

Re: [sipx-users] New to sipx - call coverage issues

Hello - I didn't get any responses to the below; I was hoping that somebody could point me in the right direction. Let me know if there are any takers. Thanks! -Ben From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of R. Benjamin Kessler

Re: [sipx-users] after update Openfire server won't start

There probably would not be "noticeable" changes for sipxecs in 1.05 to 1.06. Version 1.07 (when released) would be awesome so one could build upon t.38... The real issue is how to fix the openfire function so one does not have to do these workarounds to get it running. On Mon, Aug 2, 2010 at 2:2

Re: [sipx-users] after update Openfire server won't start

The freeswitch move to a newer version should rather be an improvement I would say ... Am 02.08.2010 um 16:43 schrieb Tran, Ly V.: > I'm showing Freeswitch ver 1.0.6 as well after the yum update. Openfire > server will not start as well. > > Ly Tran > > -Original Message- > From: sipx

Re: [sipx-users] Unsubscribe

On Mon, Aug 2, 2010 at 11:24 AM, Matthew Kitchin (public/usenet) wrote: > I think you need to try one of these methods > > List-Unsubscribe: , > > I've unsubscribed him. Th

[sipx-users] List-Unsubscribe

List-Unsubscribe: Dave Post Professional Computer Services 928-300-6548 1902 Bonita Dr. Cottonwood, AZ 86326 ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/

Re: [sipx-users] Unsubscribe

I think you need to try one of these methods List-Unsubscribe: , On 8/2/2010 9:42 AM, Dave Post wrote: unsubscribe Dave Post Network Systems Administrator 928-300-654

Re: [sipx-users] after update Openfire server won't start

On 8/2/10 11:10 AM, Douglas Hubler wrote: Freeswitch 1.0.6 must have made it into 4.2.1 somehow f just before ezuce published their update, and for about 12 hours, there was another update SHOWING (and as some have mentioned. it seems to have a higher build number than the ezuce one) maybe a

[sipx-users] Unsubscribe

unsubscribe Dave Post Network Systems Administrator 928-300-6548 928-634-8500 X7014 Genesis Worldwide Enterprises, Inc. 4600 South Genesis Drive Cottonwood, AZ 86326 ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list

Re: [sipx-users] after update Openfire server won't start

I'm showing Freeswitch ver 1.0.6 as well after the yum update. Openfire server will not start as well. Ly Tran -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Gerald Harper Sent: Monday, August 02, 2010 1:36 AM

Re: [sipx-users] Setup of DID on HA

On 8/2/2010 7:59 AM, Tony Graziano wrote: > Not necessarily. Voip.ms requires the registration to be there in order > to actually send calls out anyway. No, you can use a sip uri. Which looks like it would be better for a HA setup as it would use DNS. > So yes, you'll get calls, but sending > ca

Re: [sipx-users] Setup of DID on HA

And don’t forget you'll need to have both gateways in the dial plan entries with your preferred gateway first. On Aug 2, 2010 8:01 AM, "Tony Graziano" wrote: Not necessarily. Voip.ms requires the registration to be there in order to actually send calls out anyway. So yes, you'll get calls, but s

Re: [sipx-users] Setup of DID on HA

Not necessarily. Voip.ms requires the registration to be there in order to actually send calls out anyway. So yes, you'll get calls, but sending calls would be problematic wouldn't it? Which server has the sipXbridge role assigned to it? If you have "two" servers both with sipXbridge assigned, the

[sipx-users] Setup of DID on HA

It looks like the best way to setup inbound calls from an ITSP (voip.ms) with a HA system behind a firewall is by the calls coming in via URI? The firewall would need an hole for each server used for fallback as defined in the public DNS? That would also mean there is a sipx SBC for each HA

Re: [sipx-users] one way audio on inbound calls

Can you try with a softphone (like xlite) to see if that phone properly handles the offer from the provider? On Mon, Aug 2, 2010 at 6:14 AM, Irena Dolovčak wrote: > There is no STUN involved, i set the static ip instead. > > I have some new information from my provider. They say, the problem is

Re: [sipx-users] one way audio on inbound calls

There is no STUN involved, i set the static ip instead. I have some new information from my provider. They say, the problem is with the interoperability; they are using asterisk and the problem starts when sipx sends the ringback tone. their server doesn't register the call for 10-15 sec and it do

Re: [sipx-users] one way audio on inbound calls

Mike, thanks for your reply. I have no SIP helpers enabled.. The first thing I do is to turn them off.. So, I have contacted my provider, and they say that their server doesn't send anything when media should be sent. But i'm really not sure if that's their problem.. I think it could be that they

Re: [sipx-users] How to enable MUC conversation logging to database and this to be persistent setting for all new chat rooms?

Hi, yes i agree that it is necessary to be found a general solution and not a solution for particular cases. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/