They are working on this functionality for stable release 4.2 of
sipXbridge. Not sure if it is even ready in the development 4.1 version
yet.
To make this work now you would need a different session border
controller.
Mike
From: sipx-users-boun...@list.sipfoundry.org
Sounds like you're playing with the development version of code...
Maybe one of the Dev's has the specs that they are designing to.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Carlos Robotti
Sent: Wednesday, August 26,
Christopher,
What you might want to do is open a ticket on this
(http://track.sipfoundry.org) and possibly the person that coded the
configuration page for the Linksys phones could modify the templates...
Mike
From: sipx-users-boun...@list.sipfoundry.org
Thanks for doing some work on these phonesI know I have always been
intrigued by the linksys phone but never used them.
It would be great if you could test these features, document what needed to be
done and then we can get the wiki updated to show what features it really does
support.
I'm just downloading my ISO now and hope to be ordering this service
later in the week.
On the off chance there could be a problem with it - anyone using it?
http://www.unlimitel.ca/temp/services/voip_services/voip_ala_carte.html
I am using it and it hasn't missed a beat since I started last
On Wed, 2009-08-26 at 23:27 -0500, Christopher Coleman wrote:
The overall questions is... What is required to make a SIP to SIP
phone call over an Internet connection?
as usual, the answer is it depends :-)
For example: m...@mydomain.com wants to call y...@yourdomain.com. We both
are
Sorry for the long delay. I seem to be running firmware version 4.2
(2007-09-19) on hardware version 4.4.
I guess I'll have to risk doing a firmware upgrade to get the MoH
working. Thanks for your help.
Keith.
Picher, Michael wrote:
I meant firmware on the Patton... sorry for not
Christopher Coleman wrote:
[...]
I know I probably went into more detail than needed for your questions,
but I thought it might help.
I've also attached my current modified plugin files so you could compare
to what comes with SipX if you wanted to do something similar.
The linksys
On Thu, 2009-08-27 at 09:01 -0400, Tony Graziano wrote:
Can you route ISN calls via sipxbridge in 4.01?
ISN calls are handled just like global/Internet SIP calls -- that is,
the ISN dial string is converted into the destination SIP URI, which is
handled in the general manner. (Assuming that
Don't suppose there is a way of using MySQL instead of PostgreSQL?
I ask because I need to do a small integration between a MySQL based web setup
and SipX.
Basically, I need to have a way of synchronizing users/options between a MySQL
based web site and the SipX users/options.
I'm not looking
The sip gateway piece near the end of the config is a tad different. I
have some 5.x examples in the wiki... let me know if you need help with
it.
From: Keith Gearty [mailto:ke...@glensound.co.uk]
Sent: Thursday, August 27, 2009 9:24 AM
To: Picher, Michael
Cc:
Situation: 2 gateways set up for two different locations. One user
group is set to send calls (local and emergency) out of one gateway,
the other group out the other gateway. sipX will obey the source call
routing rules on the local dial plan rule and send the call out the
appropriate gateway, but
I attached the XML file (with some IP and name modifications to
protect the innocent :-P ) and now that I'm looking at it, the
emergency section doesn't have any callerLocationMatch tags in it.
Is sipXconfig supposed to write those out for the emergency rule?
On Thu, Aug 27, 2009 at 3:21 PM,
Situation: 2 gateways set up for two different locations. One
user group is set to send calls (local and emergency) out of
one gateway, the other group out the other gateway. sipX will
obey the source call routing rules on the local dial plan
rule and send the call out the appropriate
Does anyone know of a way of dealing with users from the command line?
For example, a CLI method of adding, removing, users, changing their
permissions, password?
Mike
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive:
OK, I manually modified my /etc/sipxpbx/fallbackrules.xml emergency
section to look like the following, restarted the proxy and registrar,
and the location based routing is now working (as a temporary
workaround(names and locations changes to protect the innocent)):
userMatch
On Thu, 2009-08-27 at 16:38 -0500, Josh Patten wrote:
OK, I manually modified my /etc/sipxpbx/fallbackrules.xml emergency
section to look like the following, restarted the proxy and registrar,
and the location based routing is now working (as a temporary
workaround(names and locations changes
I understand, I'll just have to remember to change the emergency
section each time I make a change.
On Thu, Aug 27, 2009 at 4:53 PM, Scott
Lawrencescott.lawre...@nortel.com wrote:
On Thu, 2009-08-27 at 16:38 -0500, Josh Patten wrote:
OK, I manually modified my /etc/sipxpbx/fallbackrules.xml
On Thu, Aug 27, 2009 at 8:23 AM, Keith Geartyke...@glensound.co.uk wrote:
I guess I'll have to risk doing a firmware upgrade to get the MoH working.
Thanks for your help.
You will have to rewrite the SIP portion of your config if you
upgrade. If you can wait a few days, I have a 4114 here in
I am testing MOHR among the sip phones of Polycom (BootROM- 4.1.3.0052/ App
3.1.3.0439), Grandstream (1.1.6.46) and Eyebeam (1.5.20) now. Seems only
Polycom Soundstation supports MOH with sipx.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
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