[?]
On Fri, Mar 11, 2011 at 3:10 PM, Nathaniel Watkins <
nwatk...@garrettcounty.org> wrote:
> Talked to Jim C - I was registering to the FQDN - not the SRV - once I
changed it to the SRV - the outbound caller-id was transformed correctly.
>
>
> -Original Message-
> From: sipx-users-boun...
Talked to Jim C - I was registering to the FQDN - not the SRV - once I changed
it to the SRV - the outbound caller-id was transformed correctly.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel Watkins
---
#2 - see the wiki for examples for fxs config for patton!
---
Sample Patton config did the trick.
Next part of the puzzle - it is only sending out the 4 digits for outbound
caller-id. So I changed the Subscriber Number on the FXS interface. As soon
as I change this to the full num
The reason is that the ITSP is treating a session refresh as a fresh call
(ITSP BUG).
Here is what I think is happening:
Call dies without ITSP sending BYE resulting in a stuck call.
SipxBridge sends re-INVITE (session timer) half an hour later.
ITSP treats it as a fresh call ( should reply with
I found a Jira for this problem as applied to Linksys phones. I tried setting
the config parameter noted there to disable authen for these messages, but so
far I have not gotten it to work.
But this is exactly the info I need to keep trying. I'll do some traces.
Thanks,
Jeff
On Mar 11, 201
I think if you are not satisfied with what
http://track.sipfoundry.org/browse/XX-9346 tries to do, you should
either comment on it or open an enhancement request in the JIRA and
ask people to vote for it.
On Fri, Mar 11, 2011 at 11:32 AM, Geoff Van Brunt
wrote:
>>>it's annoying that the dial-by-n
>>it's annoying that the dial-by-name switches from input to selection
when it has narrowed the field to three matches Is there a setting that
controls this? If not, is it JIRA-worthy?
I think callers expect Auto Attendants to act relatively the same no
matter who the vendor is so changing the
In a company this size, I don't think that aborting the "spelling" phase at
three matches makes it less complicated for the user.
With one more spelling button they could hit the match exactly instead of
having to wait for the AA to read out the three selections.
Listening to the AA read out th
If it makes it less complicated for the caller, why would you want to
prolong their button pushing?
On Fri, Mar 11, 2011 at 10:58 AM, Burden, Mike wrote:
> In a semi-related issue, in a very small company (like mine), it's annoying
> that the dial-by-name switches from input to selection when it
In a semi-related issue, in a very small company (like mine), it's annoying
that the dial-by-name switches from input to selection when it has narrowed the
field to three matches. In our office, that's half the company!
Is there a setting that controls this? If not, is it JIRA-worthy?
Sipx sends Notify message with check-sync header to the phone. It works only
for registered phones.
My spa942 phone was by default configured to challenge that Notify, and sipx
can not handle that challenge.
So there is a possibility, that you just have to configure your spa ata's
not to challeng
We are using SipX (4.2.1-018971.dhubler 2010-08-21T04:59:18 build34). Using
purely a siptrunk through an ITSP. CDRs does not record any of these calls.
We think it is similar to XX-6698 because in both cases, the calls were made
from two different remote sipx registered phones to two differen
sipXecs issues a NOTIFY message with the Event: element containing the value
check-sync. That is the method implemented by most IP phones.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jeff Gilmore
Sent: Friday, March 11, 2011 5:43 AM
People might want to vote on this as well. So far I seem to be the only
one. I know with our systems the #1 complaint was Dial by Name problems
back in 4.0-. It's pretty important that this work flawlessly. Important
enough that I think it should be fixed in 4.4. That won't happen without
votes...
Can anyone tell me how Sipx implements the functionality that causes phones to
reboot themselves? This feature works fine for Polycom devices, but I have a
network of Linksps SPA2102 ATA devices, and it doesn't work. I wondered if
there was some script I could alter (and submit as a patch to 4
>>> On 3/10/2011 at 08:35 PM, in message
,
"Paul Herron" wrote:
> Tony is exactly right.
>
>
>
> If you log into the Vitelity user portal, under the Support tab>Generic
> SIP subtab; Vitelity will tell you the proxy URL or IP address that you
> should be using in SipX under Gateway>ITSP Accou
XX-6698 is marked as fixed in 4.2.
You run "latest production" version? What is it? Is it 4.2.1?
Why do you think you encountered XX-6698? Did you compare traces?
Do you see those calls in sipx cdr? That is, it may happen that your ghost
calls go from "something" directly to your pstn gateway.
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