Matt White wrote:
Michael has offered a domain name he owns (which I think would work
well), but I have not heard any other feedback. But there is no rush.
I'm hoping that the integration keeps everyone happy. Personally I
will stay using the mailing list (not to knock the forum I just spe
Administrator wrote:
Well after the last round of people requesting a
forum...I've gone and created one.
Now the good thing is that it is integrated with the current
Sipx-users mailing list. So email sent to the list will be
added to the forum as a thread. And posts added to the
forum will be
Josh Patten wrote:
>I am sure this question has been asked before, but I'll ask again in
>case new information is available:
>
>Has anyone had good experience with any particular DECT/Wi-FI phones and
>sipX? Does sipXconfig have any management capabilities for any of these
>phones? (my guess is
Scott Lawrence wrote:
>The search box in the upper right of the http://sipxecs.sipfoundry.org/
>page searches both mailing list archives and the wiki all in one
>operation. Dale even worked out how to add that to the list of searches
>you can do from your Firefox search box.
>
>
I can't even fi
Picher, Michael wrote:
>There's no doubt that Trixbox is good for very small installations. It
>has a lot of functionality and can be easy to setup. For those two
>reasons alone it can make a lot of sense for home users.
>
>For those of us with corporate responsibilities sipXecs is a more robust
Picher, Michael wrote:
>We've tried them with a Mitel 3300 pbx but not with sipXecs... They are
>simple to setup.
>
What was your opinion of their quality, etc?
___
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Keith Gearty wrote:
>Has anybody tried the Patton M-ATA-1/E with SipXecs? (It's basically a
>miniture 1 x FXS gateway). We're looking at using it together with our
>analog DECT phones as an inexpensive replacement for the Snom m3, which
>was universally disliked.
>
&
Has anybody tried the Patton M-ATA-1/E with SipXecs? (It's basically a
miniture 1 x FXS gateway). We're looking at using it together with our
analog DECT phones as an inexpensive replacement for the Snom m3, which
was universally disliked.
Does anyone know how well its consultative call trans
William Otten wrote:
> Tony
> Thanks for the help but I'm still dead in the water.
>
The link that Tony just provided is excellent and I would advise you to
follow it. Gotta be easier than trying to replace sendmail in a SipXecs
system.
Keith.
___
si
William Otten wrote:
Keith Gearty
Would you provide a copy of your files: sendmail.mc, the auth file,
and whatever you feel it revilient?
Unfortunately I haven't bothered setting up mail on my current
production box as it wasn't required. I set it up on my old testing
box,
Jim Canfield wrote:
On Tue, Oct 13, 2009 at 8:28 PM, Paul Herron wrote:
I can't tell you specifically how to configure SendMail because every
installation is different. You need to pick and choose the relevant parts
for yourself. I spent about six hours reading the SendMail documentation,
Josh Patten wrote:
>Mailing lists:
>* reveal your email address to everyone that's on the list
>
>
Admittedly this is one down side of a mailing list
>* are hard to quickly glean information from
>
>
There are several mailing list syndication services which allow advanced
searchs of the mail
Dennis Wallen wrote:
>Thanks! I was trying to figure out how to trigger an SMS message off of
>emails sent to a users mailbox and had forgotten that most mobile
>providers have email to sms gateways. I can live with that solution.
>Do you know if SMS is reliable? If your phone is off will a mes
Dale Worley wrote:
>On Mon, 2009-10-12 at 06:46 -0700, Charles Chalekson wrote:
>
>
>>I have a small pbx network with three dedicated voice lines all
>>through a Patton gateway to my phone provider. I initially had local
>>dial plans where you had to dial a 9. I realized after some time that
Dale Worley wrote:
>On Sun, 2009-10-11 at 23:06 -0400, William Otten wrote:
>
>
>>Sendmail isn't that user friendly, especially if your not linux friendly
>>or a programmer. Can postfix been installed in place of sendmail? I
>>have used postfix before with trixbox, it is much easier to use. A
lly need a method of easily automating the setup.
Keith Gearty wrote:
I understand what William is asking for, as I had to do this myself to
get SipXecs sendmail working for me. I used a combination of the
SendMail options
Smart Host, Generics Table, and SMTP-AUTH.
Here are a few useful link
p://thread.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/13672
Note: You'll need to use vi to edit the sendmail configs on the SipXecs
box. vi isn't very straight forward to use, so I recommend googling it.
Kind Regards
Keith Gearty
Design Engineer
*Glensound Electron
The way I understand it (based on his previous post), he has a LAN which
currently has no internal DNS. The gateway router also acts as DHCP
server and external DNS proxy. The IP addresses of his ISP's DNS
servers are automatically configured in the router via the DSL link.
This is a very co
I think your best bet would be to simply replace the system with a
modern installation of SipXecs. The latest stable ISO version (4.0.2)
would be the one you want. Unlike SCS, SipXecs is professional quality
open source freeware. You just download the ISO file, burn it to CD,
put it into the
IT Services wrote:
>Hi there:
>
>I am using 4.0.2 with a Patton 4114 PSTN gateway. I used the config as
>suggested with the gateway forwarding incoming PSTN calls to the auto
>attendant at extension 100.
>
>I want all incoming calls to go to extension 301 (receptionist) but when
>I change the exte
Matt White wrote:
True, Avaya has no pure "softswitch" product. So it would fill a hole
for them.
My concern is just that opensource and avaya are not the best of
friends. And with SipX being license LGPL they could start pulling
all new features in as non-opensource components. Leaving t
Simon Stockdale wrote:
>The reason behind pursuing the VMWare route is simply that I would like to
>deploy sipx to support more than one customer. Our options at present are
>either modify sipxconfig code to provide some grouping of users to specific
>companies or deploy multiple instances of SIPX
Jim Canfield wrote:
On Thu, Aug 27, 2009 at 8:23 AM, Keith Gearty 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 h
r not clarifying...
*From:* sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Picher,
Michael
*Sent:* Friday, August 21, 2009 1:18 PM
*To:* Keith Gearty; sipx-users@list.sipfoundry.org
*Subject:* Re: [sipx-users] Getting Music On Hold (MoH) to
Hi all
I know I've asked something like this once before, but I didn't get a
reply that helped me resolve the issue.
I have a Patton SmartNode 4114 PSTN gateway with 4 FXO ports, and I
would like to get Music on Hold working through it. My internal phones
are Snom 320s, and they have a "Mus
Damian Krzeminski wrote:
>In many places sipXconfig is using a validator that accepts either "dial
>digits" or "sip URIs". I am pretty sure that it can/should be relaxed.
>Please open an issue.
>D.
>
Done. http://track.sipfoundry.org/browse/XX-6358 I wasn't sure which
component to put it under,
Picher, Michael wrote:
> I don’t think you can… see my previous response.
>
> Set an if-no-answer forward option to the wireless phone. In the phone
> configuration look for the call forward on busy option and make that
> dial 8+ext to forward immediately to VM.
>
> I’m fairly certain that shoul
Keith Gearty wrote:
>I just found an annoying little bug under User > Call Forwarding. The
>"forward to" field is supposed to accept any potentially valid SIP
>extension, but it seems to only accept numeric extensions or fully
>qualified SIP URIs. If you put i
jun,wen wrote:
Hi, Team,
I am trying to implement a blacklist to users to bar some specific
call destination. when created a permission of "Blacklist
Access", I made a new dial plan of "Blacklist" with specific barred
PSTN number by a prefix and checked the "Blacklist Access" as the
requi
I just found an annoying little bug under User > Call Forwarding. The
"forward to" field is supposed to accept any potentially valid SIP
extension, but it seems to only accept numeric extensions or fully
qualified SIP URIs. If you put in a non-numeric extension without the
SIP domain part, it
Scott Lawrence wrote:
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
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
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.
___
sipx-users mailing list sipx
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
Dennis Wallen wrote:
>I have a Snom 320 and just configured function key P1 as you described
>below (12 digits total). After the phone rebooted I checked the phones
>web interface and the speed dial was there as expected. I don't that
>this helps other than to know that it should work. Let me k
I'm trying to set up phone-based speed dialling on my phones (Snom 320)
using the SipXecs web interface and auto-provisioning. On the Snom 320
> Function Key page I set function key P1 to type Speed Dial then put
the number in (which is 9 + <11 digit number>). I then get an error
message "Nu
Dan White wrote:
I need to change the IP address of the sipx server, and the gateway,
where can that be done and if its unix how?
Please see here: http://list.sipfoundry.org/archive/sipx-users/msg16264.html
and here: http://list.sipfoundry.org/archive/sipx-users/msg16289.html
Keith.
1. Edit the file /etc/sysconfig/networking-scripts/ifcfg-eth0 (assuming
eth0 is your Ethernet card)
2. Restart network services using service network restart
3. In the web interface under 'Servers', change the expected IP address
of the Primary Server, send all profiles and restart affected
Todd Hodgen wrote:
Dale Worley wrote:
>A particular problem is that in order to enforce a
>bandwidth restriction, the SDP editing must remove any codecs that the
>filter does not understand. This automatically interferes with
>endpoints introducing new codecs.
>
>
Wrong. We're talking about an
Scott Lawrence wrote:
On Fri, 2009-08-07 at 09:37 +0100, Keith Gearty wrote:
Paul Mossman wrote:
The admin should see simply a list of codecs that can be selected and
de-selected.
As I understand it, SipXecs cannot and should not work quite like that.
As far as SipXecs
Paul Mossman wrote:
>The admin should see simply a list of codecs that can be selected and
>de-selected.
>
As I understand it, SipXecs cannot and should not work quite like that.
As far as SipXecs cares, the codec names are nothing more than text
strings. RTP packets can carry streaming medi
Scott Lawrence wrote:
>As long as any call to the PSTN requires some permission, an attacker
>needs to be able to guess the password of a user with that permission.
>
Out of interest, is there any brute-force detection & lock-out in SipXecs?
Keith.
___
Scott Lawrence wrote:
>The difficulty is, I think, that the auto-attendant will not transfer a
>call to something that is not an 'extension' (either a user or an alias
>for a user).
>
Surely that's intended behaviour. Internal dial rules and prefixes are
normally considered privileged functions.
Different versions of firmware may have different features enabled
requiring a different amount of power. It is quite possible that
changing to a different version of firmware may cause these POE problems
to surface. As you are running on POE, you should do as Jim suggested
and change to exte
Scott Lawrence wrote:
From: naga raju
is sipx 4.0 support skype trunking?
Picher, Michael wrote:
No.
That is correct in that sipXecs does not implement the current Skype
phone protocol, but Skype is introducing a new service to connect PBX
systems to Skype using SIP. We
Mark Eissler wrote:
>Specifically, when configuring an Auto-Attendant under "Actions" the
>following is stated: "To allow the attendant user to dial someone's
>extension at any time, do not assign an action to the first digit of the
>digits in the range of internal extensions. For example, if y
Nitin Mirchandani wrote:
On a user list - I am looking for onstructive ideas as I feel the
project I have is quite complex(atleast for me) and I will be ditching
trix for sure as its a admin hell.
Sipx seems clean and simple but as I have nil experience, need "step
by step" help.
Ok well her
Scott Lawrence wrote:
No... he said they were connected by a VPN. You're right that you
cannot use HA if there is a NAT between the systems.
Oops, I missed that.
But I don't know of any way to configure the phones to get the effect
he asked for - that they use HD locally and a more economic
Scott Lawrence wrote:
Nitin Mirchandani wrote:
Now - Help required in
a) Interconnecting SipXA to SipXB
The easiest way to do this is to just make the two systems an HA
cluster; both will be managed together as a single system with a unified
dial plan. If either fails, the other take
naga raju wrote:
Hi all,
I am installed sipx 4.0 successfully. Then i tried to add some users
by using web ui. After clicking apply button by entering user details
i am getting error like " An internal /error/ has occurred please
click here to continue". if i click the continue button it's lo
Dale Worley wrote:
On Fri, 2009-07-17 at 13:58 +0100, Keith Gearty wrote:
The comments at the bottom of the article (by a Microsoft guy who was
working on Response Point) seem to deny the claims in the article.
All I see are notices that people are leaving the project and that the
The comments at the bottom of the article (by a Microsoft guy who was
working on Response Point) seem to deny the claims in the article.
___
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Hello Rene
Please see here:
http://sipx-wiki.calivia.com/index.php/Create_a_wiki_account
Keith.
Rene Pankratz wrote:
I would like to do that but I cannot create an account for the wiki.
It seemsthe links are broken there...
http://sipx-wiki.calivia.com/index.php/Help:Contents
When I cli
Tony Graziano wrote:
> I've always been under the impression you should only call a line in a
> hunt group ONE TIME.
>
> Provisioning a second line on the recptionista phone and ringing THAT
> line might be a better workaround.
I've never had a problem with having a phone mentioned twice on a h
I can't find a MOH server setting anywhere in the SmartNode web
interface. Does anyone know if MOH is supported on the Patton SmartNodes?
Regards,
Keith.
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List Archive: http://list.sipfoundry.or
You don't need to use two hunt groups to do this. You can do it with
just one:
First... Ring receptionist, timeout: 6 seconds
If no reply... Ring receptionist
At the same time... Ring other person 1
At the same time... Ring other person 2
At the same time... Ring other pers
the primary server as the configuration master.
Keith Gearty wrote:
Its tricky for sure. SipXconfig on the secondary server could become
active after a timeout of no contact with the primary server. You
then use an rsync based process when they find each other again.
Keith.
Scott Lawrence
nions on which services that are not currently
redundant that users thing should be. If you could pick just one
service for us to add HA to, what would it be?
On Tue, 2009-07-14 at 14:41 +0100, Keith Gearty wrote:
SipXconfig. The ability to be able to administer the system through
th
I believe there's a CSV import function that may suit your purposes.
Trying creating a new test users then export a CSV file to see what the
exact file format is.
Keith.
jun,wen wrote:
Hi, is there any tools to create subscribers in bulk in sipx ?
Regards
Jun
-
Scott Lawrence wrote:
On Tue, 2009-07-14 at 10:05 +0100, Keith Gearty wrote:
Basically from what I can gather, the intended purpose of HA systems
is load balancing, not true redundancy. It would be nice if it were
designed with both purposes in mind, but it doesn't seem to be.
Basically from what I can gather, the intended purpose of HA systems is
load balancing, not true redundancy. It would be nice if it were
designed with both purposes in mind, but it doesn't seem to be. In my
system I've implemented a simplistic redundancy by cloning the SipXecs
server and leav
heros wrote:
More I see another workaround.
1) Create a permission no_customer_1 à assign this permission
to customer 2 with extension 500
2) Create a permission no_customer_2 à assign this permission to
customer1 with extension 600
3) Create a rule in dialplan with p
oesn't have that functionality then its something that
really should be added.
Keith.
Sven Evensen wrote:
>If I want to redial, I would need 90047.
>But our application just needs to determine if incoming call is national
>or international.
>
>Sven
>
>-
Todd Hodgen wrote:
I don't see a method of having a centralized receptionist for this
application. Many times a multi-tenant application will be for shared
resources - like receptionist. Like Fax machines, etc.
One other aspect of Multi-tenant applications is a common use of
trunks. In
Sven Evensen wrote:
> We are in UK and have a SIP trunk to an ITSP in Norway. When calls
> come in from the Norwegian trunk from Norwegian
>
> numbers (ie the calling number), they do not have the country code
> prefixed, so I get 12345678 while I want 4712345678. The ITSP say they
>
> cannot pr
You could set up extension forwarding so that the call gets forwarded to
someone else if the callee is busy, or after a certain number of rings.
Alternatively, you could set up a queue so that the external caller gets
put "into a queue", and will then be taken out of the queue when the
callee
I had this headache as well a month or so ago. Here's a quote from a post back
then on how to sort it:
>if you installed on Centos
>edit
>
>/etc/sysconfig/networking-scripts/ifcfg-eth0
>
>(assuming eth0 is your Ethernet card)
>
>then restart network services
>
>service network restart
Regards,
natif wrote:
>but even with patton (MP4114)
>I've some issue (call not hangup when called hangup, call cut...).
>Issue are send to the support of Patton, waiting for an answer (but
>support tell me that they are working on...).
>
Here are the 2 main issues I encountered with the Patton gateways,
+1 Agreed.
The fact that you can't select a path on the FTP server in which to
upload the backup is crippling IMO.
Keith.
Matt Keys wrote:
>>From the web interface: System -> Backup, then specify FTP. I get a
>IP/Hostname box, username, password. If I enter just the IP or hostname
>it'll work,
Tony Graziano wrote:
>
> Patton is good and supported worldwide.
The Patton SmartNodes are very powerful and provide a lot of advanced
features, but be warned that they have a VERY steep learning curve, and
can be very daunting for a new user. Basically they are supplied in a
non-working stat
heros wrote:
>>But how can that be, since PSTN calls don't send a Called Party ID?
>>Surely you cannot retrieve the DID number from a PSTN call.
>>
>>
>
>
>
>>Keith.
>>
>>
>
>
>With a sniffer or tcpdump look at the "TO:" field in the SIP INVITE.
>Example: TO: 0122...@mysipdomain.it
>T
But how can that be, since PSTN calls don't send a Called Party ID?
Surely you cannot retrieve the DID number from a PSTN call.
Keith.
heros wrote:
>SipX route the calls from a sip PSTN or GSM gateway using the rules in
>dialplan. If no rule that match incoming DID is defined than SipX rejects
Since you mention PSTN and GSM calls, I assume you are using an external
gateway, rather than SipXbridge. As far as I'm aware (would someone
please correct me if I'm wrong) SipXbridge can only be used for calls to
and from an ITSP. I use a PSTN gateway, and I set it up to modify the
To: field
As others have said, this is a phone setting not specifically a SipXecs
setting, because the beep is generated by the phone not by SipXecs. If
you are using auto-provisioning to distribute the phone settings from
the SipXecs server to the phones, then this may be adjustable via
"Phones" in Sip
SipXecs is open source under the GNU General Public Licence, so even if
the worst happens the project can still be continued by anyone who has
downloaded a copy of the source code.
Keith.
Goran Donev wrote:
> As we all have been reading it’s bad news for Nortel.
>
> The question I have is wit
When you manually added the phone, did you enter the correct case for
the MAC address?
Keith.
Boy Aidil Sjam wrote:
>Oops, I mean, I found spa000E08D09436.cfg at the tftproot folder. Sorry for
>the typo
>
>
>-
>Original Message:
>From: Keith G
Was that a typo in your email, or does the file actually have the wrong
number of zeros after "spa" ?
Keith.
Boy Aidil Sjam wrote:
> Hi All,
>
> I think there are some issues with Linksys SPA IP Phone with sipXecs
> ver.4.0.
>
> I used Linksys SPA941
>
> First, when I used Discover Devices, th
Damian Krzeminski wrote:
> If you use stable repos (with 'pub' in the URL) it'll bring your software
> to stable release.
> If you use unstable repos (with 'temp' in the URL) it'll bring your
> software to developement release.
> It works in the same way as with Fedora repos.
> D.
>
>
Am I right
I just tried disabling the two AVG 8 Firefox extensions, and it didn't
make any difference.
Mark Howells wrote:
>
>
>-Original Message-
>
>
>>From: Keith Gearty [mailto:ke...@glensound.co.uk]
>>Sent: 18 June 2009 11:01
>>To: Mark Howell
I figured it out. My wav file has to actually be called default.wav in
order to work. Thanks for your help.
Keith.
Keith Gearty wrote:
>Sorry for confusion. I'm using 4.0.0, trying to upload a new user
>voicemail greeting (/"Hello, I'm not in the office right now, bla
0.0.
>-----Original Message-
>From: Keith Gearty
>To: Tony Graziano
>Cc:
>
>Sent: 6/18/2009 10:04:45 AM
>Subject: Re: [sipx-users] Uploading a voicemail greeting
>
>
>Tony Graziano wrote:
>
>
>
>>>>>
Tony Graziano wrote:
>>>>On 6/18/2009 at 9:40 AM, in message <4a3a43bb.3050...@glensound.co.uk>,
>>>>Keith
>>>>
>>>>
>Gearty wrote:
>
>
>>Tony Graziano wrote:
>>
>>
>>
>
Tony Graziano wrote:
>>>>On 6/18/2009 at 9:28 AM, in message <4a3a40f0.4060...@glensound.co.uk>,
>>>>Keith
>>>>
>>>>
>Gearty wrote:
>
>
>>Maybe I'm just missing something obvious, but I can't find a wa
the same way that I do for the auto attendant greeting and the music on
hold. Where is the upload option for voicemail greetings?
--
Kind Regards
Keith Gearty
Design Engineer
*Glensound Electronics Ltd*. 1,5 & 6 Brooks Place, Maidstone, Kent, ME14
1HE. UK
Tel: +44 (0) 1622 7530
I have similar problems accessing the web UI through Firefox, and have
had to resort to accessing it through IE. Are you using Firefox? Do
you also find that the "Show Advanced Options" links don't work? I
posted about this issue a while ago, but apparently no one else was
seeing those probl
If I understand your situation correctly, then this can all be done with
a single hunt group, without the need for call forwarding or second
lines
First ring receptionist, timeout 10 seconds
If no answer, ring receptionist
At the same time, ring someone else in hunt group
At the same time, r
Yep it all seems to be working again. Todd just made a typo in the
subject line when he said sipfoundry.com. He meant sipfoundry.org.
Kind Regards
Keith Gearty
Design Engineer
*Glensound Electronics Ltd*. 1,5 & 6 Brooks Place, Maidstone, Kent, ME14
1HE. UK
Tel: +44 (0) 1622 75
Confirmed. Nothing on the sipfoundry.org server seems to be accessible
(including the links you provided). The SipXecs Project server at
sipxecs.sipfoundry.org appears to be working fine. I guess its either
known maintenance, or another DNS caching issue if the site has just
been moved to anoth
Sorry to butt in here.
I know a bit about VoIP, but not much about Fax over IP. I'd be curious
to know why Error Correction (ECO ??) doesn't work when the data stream
is routed via IP. I assume we're talking about POTS fax machines
connected via an FXS->IP gateway? Or are these actual IP fax
You have to set up a phonebook in Features > Phonebooks from the
superadmin interface first. You select a User Group of users to be
included in the phonebook, and a User Group of users who are allowed to
see the phone book.
Regards,
Keith.
Gabor Paller wrote:
> Hi,
>
> I have an extremely si
Directed Call Pickup can be configured from System > Servers >
[server-name] > SIP Registrar. The default pickup code is *78
Yakout Esmat wrote:
>I can't even find a way to configure it
>
>-Original Message-
>From: Dale Worley [mailto:dwor...@nortel.com]
>Sent: Thursday, 4 June 200
I'm setting up a weekly backup job on my SipX server which is completely
seperate from sipxconfig. It already backs up all the major Linux
config files, but I just wanted to check with you guys which SipX
specific directories ought to be included as well. The way I see it, I
should be includi
I notice that the default MOH music has changed in 4.0, and is now a
real piece of music that isn't painful to listen to. What is the
license on this default music? Is it free from all requirements for
performing rights licenses and the like?
___
s
Ok I found out about the command /sipxconfig.sh --database
reset-superadmin/. That set my superadmin pin to blank and allowed me
to log in. But now when I try to change the superadmin pin from the web
interface it says "An internal error has occurred" then logs me out.
Keith Ge
I seem to have accidently changed the superadmin pin, but I'm not sure
what to. Is there any way I can reset it using the console?
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h the config files inside the configuration backup, I can
see lots of references to the old domain name. I suppose I should go
though each of those and rename them to the new domain name?
Scott Lawrence wrote:
>On Tue, 2009-05-26 at 17:58 +0100, Keith Gearty wrote:
>
>
&
I did a clean installation of SipX 4.0.0 and restored a backup of my
previous 4.0.0 system (exact same build). The only difference between
the two is that the new installation has different DNS settings. I
realised that I needed to go into System>Servers and change the hostname
of the primary
If you choose to import the users/phones from CSV, please note that it
will not import the user groups and phone groups. The groups will be
re-created, but with default settings. This was a source of musch
confusion for me in the past.
Keith.
Tony Graziano wrote:
>I'm not sure I would agr
In 3.10 the directed call pickup code could be changed in System >
General > Call Pickup. In 4.0 that menu location doesn't exist and I
can't find the directed call pickup code anywhere. Where has it moved
to in 4.0?
Thanks,
Keith.
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