On Wed, 2009-08-19 at 16:24 +0100, Keith Gearty wrote:
> I have a need to limit the number of incoming calls on a particular line
> to 1. So if that user is already handling an incoming call, any further
> incoming calls go straight to voicemail or next user on hunt group
> (instead of ringing
Keith,
What if you setup that wireless phone on a separate extension and then
have a forwarding rule on the primary extension.
Then it will ring the wireless if nobody answers the primary line.
Then in the Polycom phone configuration there is a place where you can
call forward busy and set that
Tony Graziano wrote:
>I "think" this is a function of the phone.
>Ex: Polycom
>Device>Call Handling>
>
Thanks Tony, but a phone function won't help me here, because I have two
different phones, each of a different model, on the same line. Once one
of the phones picks the call up, I don't believ
Damian isn't asking for a screenshot, he's asking for a "sipx-snapshot",
which is a built-in tool that takes a lot of different log files from
the server and compiles them together into a single XML file. You can
run sipx-snapshot by going to Diagnostics>Snapshot in the menu system,
then downl
I "think" this is a function of the phone.
Ex: Polycom
Device>Call Handling>
>>> Keith Gearty 08/19/09 11:25 AM >>>
I have a need to limit the number of incoming calls on a particular line
to 1. So if that user is already handling an incoming call, any further
incoming calls go straight to v
Anti-Virus, no, none on ff.
Mike
On Wed, 19 Aug 2009 16:34:39 +0100, Keith Gearty wrote:
> Are you using Firefox with the AVG 8 extensions? Some users have
>
> reported problems with the SipXconfig web UI under these conditions. If
> this might apply to you, try using IE or disabling the AVG
I had restarted the server before I posted that. Can I ask what the screen shot
is for considering this is such a basic error?
On Wed, 19 Aug 2009 11:34:12 -0400, Damian Krzeminski wrote:
> li...@grounded.net wrote:
>
>> For what it's worth, I've entered an issue.
>>
>
> http://track.sipfoun
Are you using Firefox with the AVG 8 extensions? Some users have
reported problems with the SipXconfig web UI under these conditions. If
this might apply to you, try using IE or disabling the AVG 8 extensions.
Keith.
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li...@grounded.net wrote:
> For what it's worth, I've entered an issue.
>
http://track.sipfoundry.org/browse/XX-6350 ?
Sorry but without sipx-snapshot it's really a lost cause.
If you can get it someone we'll have a look. If not it's probably will get
closed until someone else sees the same thi
I have a need to limit the number of incoming calls on a particular line
to 1. So if that user is already handling an incoming call, any further
incoming calls go straight to voicemail or next user on hunt group
(instead of ringing the wireless phone on his belt, which is on the same
line).
H
For what it's worth, I've entered an issue.
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li...@grounded.net wrote:
>> We do need a snapshot to investigate in such cases. If you were to
>> open a JIRA issue and attach a snapshot taken right after you see that
>> it would help.
>
> What is the correct URL for this report site? I'm finding a lot of
> things on google but nothing very spe
> We do need a snapshot to investigate in such cases.
> If you were to open a JIRA issue and attach a snapshot taken right after
> you see that it would help.
What is the correct URL for this report site? I'm finding a lot of things on
google but nothing very specific.
I just got another such err
PS: In this case, this is all there was so there's not much to send. I just hit
Click here and I was sent back to login screen.
Mike
On Wed, 19 Aug 2009 10:39:10 -0400, Damian Krzeminski wrote:
> li...@grounded.net wrote:
>
>> I don't know if I should bother reporting page errors so please let
I'll make a note of it, thanks.
On Wed, 19 Aug 2009 10:39:10 -0400, Damian Krzeminski wrote:
> li...@grounded.net wrote:
>
>> I don't know if I should bother reporting page errors so please let me
>> know.
>>
>> Here's another of these occasional errors, this time, when I tried to
>> restart th
Ah, yes, I see that the ISO does in fact put 127.0.0.1 as the first DNS server
by default.
I'll keep that in mind when using the ISO, thanks.
Mike
On Wed, 19 Aug 2009 08:37:41 -0400, Picher, Michael wrote:
> Is your PBX pointing to itself for DNS? (check /etc/resolv.conf).
>
> -Original
li...@grounded.net wrote:
> I don't know if I should bother reporting page errors so please let me know.
>
> Here's another of these occasional errors, this time, when I tried to restart
> the services.
>
> The URL shown was;
>
> https://sip.mydomain.com:8443/sipxconfig/admin/commserver/EditLoc
Hi Daniel,
I suggest you call a SipX-attached phone from OCS 2007 first. Simply create a
SipX User and use, for example, X-Lite to connect. Specify a number from your
internal number pool as the alias for the SipX user; use a number that is not
assigned to any OCS user or contact object. Yo
If it is pointing to itself, there is a known bug in 4.01 that improperly
empties the DNS zone when making changes to a gateway...
http://track.sipfoundry.org/browse/XX-6010
If you are using DNS on the system itself, check your ZONE file for the domain
sipxecs is on.
>>> "Picher, Michael" 08/19
Is your PBX pointing to itself for DNS? (check /etc/resolv.conf).
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
li...@grounded.net
Sent: Tuesday, August 18, 2009 11:36 PM
To: sipx-users
Subject: [sipx-users] DN
Is OCS setup as a gateway on the sipX box?
Does that gateway access the OCS box by name or by IP? Your gateway definition
should match how the calls are coming from the OCS system.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sip
Yes, that looks like a great solution! I hadn't thought of creating a
time based dial plan entry.
In previous versions if you just added a dial plan entry like this in
the dial plan you would never be able to dial those extensions without
having that permission. In fact, if there was another
Finally I made the tests of TFTP and FTP well completed by following steps.
For TFTP test, seems the test file of 00D01EFE is stored in
/usr/local/sipx/var/sipxdata/configserver/phone/profile/tftproot but the
original TFTP server root directory is /tftproot. I changed the root
directory to the
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