I'm not much help here - we are linking to an old NEC (not voip aware) - so we
are just routing calls via the PRI interface on the Patton.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, September 01, 2011 5:
Thanks for the log. I saw [CRIT] switch_time.c:799 Virtual Migration
Detected! Syncing Clock. A quick google for the error message yielded
this thread
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/064511.html
Anthony M. seems to be not surprized by freeswitch crashing/
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To answer your most recent question first, All 4 digit
calls match a custom dialplan, I've named 'Nortel Loca
ha ha stupid auto correct. motel should be nortel.
what type of dial plan rule are you using? what type of gateway?
On Sep 1, 2011 6:47 PM, "Tony Graziano"
wrote:
> I have found them to be wrong on occasions...
>
> if the destination of 4495 is on the motel side, why are we dialling or
> strippin
I have found them to be wrong on occasions...
if the destination of 4495 is on the motel side, why are we dialling or
stripping digits instead if defining the destination a d sending it as is?
On Sep 1, 2011 6:40 PM, "Steve Beaudry" wrote:
>
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Thanks Tony,
Absolutely nothing would make me happier than to have
someone point out a simple solution to t
to be more specific, the patton should have the dialplan entry to forward
that call to the nortel. it is referring the call back to sipx becaase it's
(patton's) dialplan is telling it to do so. i think this is really a patton
config issue.
Nate you around to field this? You did this same thing a w
what you never said was that you were trunking a nortel option 61c to sipx
via a smartnode.
I think this has been done successfully a few times. suffice it to say your
patton gateway needs a dialplan entry changed and why would you be dialling
it via a sip domain when you should be dialling it num
In Line 439 of the attached log, you can see the REFER-TO
arriving at the Patton Smartnode 4960 gateway... the
REFER-TO is destined for mailto:'4...@voip.royalroads.ca'.
This results in the gateway trying to send the INVITE back
to the SipXecs server, instead of to the correct remote
destination (
I have not either though it is a requirement to have proper dns setup at
both ends. you mentioned refer which makes me think that perhaps 1 of the
systems you are connecting with is not capable of handling refer.
explaining your call flow, gateways and user agent involved would be a good
idea.
On
I've not, what phone/softphone are you having trouble with?
On Thu, Sep 1, 2011 at 1:03 PM, Steve Beaudry
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I'm not hearing anything about this, so I'm wondering if I'm
the only one having the problem, and it's just som
i don't have an issue with you opening an issue, i'd just clearly define
what the problem is because from your description I still don't understand.
thanks,
mike
On Thu, Sep 1, 2011 at 12:25 PM, Kyle Haefner wrote:
> Mike,
>
> When a user imports contacts from google into their phone book, thi
Thanks, tracker item created:
http://track.sipfoundry.org/browse/XX-9834
On Thu, Sep 1, 2011 at 10:12 AM, Josh Patten wrote:
> This would also speed up deployments as "group speeddials" don't take into
> account that no one needs to monitor themselves with BLF so you have to
> enter a different
Mike,
When a user imports contacts from google into their phone book, this
is what what I was getting at with an outlook import.I'll create a
JIRA unless you see an issue.
Kyle
On Wed, Aug 31, 2011 at 11:46 AM, Michael Picher wrote:
> Because admins should control generating the phone books
Well I guess I will chime in. I also have a spectralink 8440 that I need to
configure, glad to know that it works. Thanks for breaking the ice. I had
previously tested the 3.3 stack on the 335 and 560 phones without issue.
--
Paul
From: Josh Patten mailto:jpat...@ezuce.com>>
Reply-To: Disc
Huzzah!!! Finally a decent wifi phone!!
On Wed, Aug 31, 2011 at 4:14 PM, Matthew Kitchin (public/usenet) <
mkitchin.pub...@gmail.com> wrote:
> This is a fun 1 way conversation, but I will continue on.
> The 8440 works great. You can tell it is built on the Soundpoint SIP stack.
> It just works.
This would also speed up deployments as "group speeddials" don't take into
account that no one needs to monitor themselves with BLF so you have to
enter a different list of users for every person in a group no matter what.
A good feature in my opinion.
On Thu, Sep 1, 2011 at 4:25 AM, Douglas Hubl
Please send me the freeswitch log offlist.
On 09/01/2011 06:54 PM, Roman Gelfand wrote:
After applying the latest update, I find that every couple of weeks
the Karoo Bridge stop working. Not sure of the reason, but freeswitch
log shows critical errors. After rebooting it works fine.
Thanks i
After applying the latest update, I find that every couple of weeks the
Karoo Bridge stop working. Not sure of the reason, but freeswitch log shows
critical errors. After rebooting it works fine.
Thanks in advance
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sipx-users
On Thu, Sep 1, 2011 at 12:29 AM, Tony Graziano
wrote:
> Users can have personal speeddials. What I am seeing is there is not
> import/export utility for the user.
contact manager for end user that doesn't allow import/export does
seem inflexible. Feature makes sense to me. So Jason would export
MAPI is really for applications generating email and connecting to it,
really to call a messaging server to send a message. Its a library to
enhance the dataset on the more commercial aspect of mail (that's why when
you install one of the big commercial email clients your mapi32.dll, etc.,
gets exc
In a lot of the use cases, someone inside who already had an account (with
their own speeddials) took over jason's duties. Either way, there is no way
to combine them except typing.
On Thu, Sep 1, 2011 at 3:21 AM, Michael Picher wrote:
> Well, wouldn't the easy thing to do be to re-purpose Jason
Well, wouldn't the easy thing to do be to re-purpose Jason's account?
Alternatively, something that would be really nice when setting up new users
would be a 'copy' command. This would let you copy settings from one user
to another (things you might not do in a group).
Mike
On Thu, Sep 1, 2011
davmail isn't really applicable. That's for multiple client types hitting
Exchange.
What most really are looking for here is a means to synchronize voicemail
from the Voicemail server in sipXecs to Exchange. The back-end integration
would allow users to listen / delete their mail in exchange and
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