This is a test message
--
David Grazio
VP of Product Marketing
o. 978.296.1005 x2016
dgra...@ezuce.com
www.ezuce.com
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Any ideas on how to get the Cisco 7961G phone working with sipXecs? I have
it working on an Asterisk installation, but when registered on the sipXecs
server, whatever I do to the phone config, I can dial voicemail (101) but
not any extensions. When dialing voicemail, I get the standard automated
we
can get approval to replace our legacy PBX, and we're probably not going to
be able to return the Ciscos, so I'd like to get them to work if at all
possible.
David Fulton-Howard
McDonogh School Technology Department
Sent from a mobile device; please excuse typos.
On Nov 4, 2012 6:18 PM, Josh
On one of our two sipXecs servers, the sipXproxy process is generating huge log
files (over 300MB/hour). Several of the same messages appear repeatedly, but
we have not been able to identify the source of the problem. Is it a bad
config file somewhere? Is it a phone that has gone crazy?
...@mail.gmail.com,
Michael Picher mpic...@ezuce.com wrote:
Hi David,
It does look like a client type situation.
2012-05-04T18:27:26.088500Z:184851336:SIP:ERR:sip1.cedarville.edu:SipClientTcp-20:41EDF940:SipXProxy:Url::parseString
no valid host found at char 0 in ';tag=6E9B7R', uriForm = name-addr
2012-05
to that.
--
---
David Becker
IANT- APPLIED NGN-TECHNOLOGIES
Schlüsselfertige VoIP-Lösungen und mehr...
IANT GmbH
Salzdahlumer Straße 46/48
D-38302 Wolfenbüttel
Fon: +49/(0)5331/ 900989-450
Fax: +49/(0)5331/ 900989-499
Internet: www.iant.de
Ust.-IdNr: DE264352710
HRB 201710, Amtsgericht Braunschweig
throw an error if the certificate is
the problem.
--
---
David Becker
IANT- APPLIED NGN-TECHNOLOGIES
Schlüsselfertige VoIP-Lösungen und mehr...
IANT GmbH
Salzdahlumer Straße 46/48
D-38302 Wolfenbüttel
Fon: +49/(0)5331/ 900989-450
an
audio file at the given URL. If the plugin itself was wasting time then
the INVITE wouldn't make it to the phone at all.
I can't reproduce the issue here so far, I'll try more.
--
---
David Becker
IANT- APPLIED NGN
.
The tones are defined in the Sound Effects category, you may have to
enable advanced settings. The internal ring tone defined there is only
used if the proxy is set to tag internal calls.
--
---
David Becker
IANT- APPLIED NGN
Hi All,
This is my first post to the sipx-users mailing list. I currently
administer a NEC PBX (NEAX 2400 IPX / SV7000) and an NEC voicemail server
(NEAXMail AD-64). The PBX communicates with the voicemail server via NEC's
Message Center Interface (MCI) using a serial cable (which we've
(latest Git?) and which build
method do you use (EDE or the new versatile makefile)? For the latest
versions CentOS is the recommended OS, not Fedora.
--
---
David Becker
IANT- APPLIED NGN-TECHNOLOGIES
Schlüsselfertige VoIP-Lösungen und
.
--
---
David Becker
IANT- APPLIED NGN-TECHNOLOGIES
Schlüsselfertige VoIP-Lösungen und mehr...
IANT GmbH
Salzdahlumer Straße 46/48
D-38302 Wolfenbüttel
Fon: +49/(0)5331/ 900989-450
Fax: +49/(0)5331/ 900989-499
Internet: www.iant.de
Ust.-IdNr: DE264352710
HRB 201710, Amtsgericht
The old EDE used to set up things like FTP, TFTP, DNS, DHCP and so on so
that everything worked as well as an ISO install but now that that's
obsolete all I know is grabbing the dependencies and then letting the
makefiles and sipxecs-setup-system do their magic, that doesn't seem to
set up the
Hi,
Can someone recommended me an IP phone that works great with Sipx? I have been
using Cisco phones for my remote users but there is always a problem, specially
with NATting..
Thanks
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List
phone for sipXecs
I've had 0 problems with Polycom IP 560's...but I've only been using them for a
week with SipXecs.
--- On Fri, 12/10/10, David Sharafy da...@ccds.ca wrote:
From: David Sharafy da...@ccds.ca
Subject: [sipx-users] recommended IP phone for sipXecs
To: sipx-users@list.sipfoundry.org
heard Snom has a decent lineup as well.
I've tried Grandstream phones with sipX and they have a lot of issues,
especially with BLF. They also have strange REFER issues.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of David Sharafy
Sent
. :( Apparently, you are saying that fails. :(
From:
sipx-users-boun...@list.sipfoundry.orgmailto:sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.orgmailto:sipx-users-boun...@list.sipfoundry.org]
On Behalf Of David Sharafy
Sent: Wednesday, November 24, 2010 2:09 PM
of sounds pretty screwed up to begin with.
On Tue, Nov 23, 2010 at 12:16 PM, David Sharafy
da...@ccds.camailto:da...@ccds.ca wrote:
thanks.. I setup the custom plan but that created issues with inbound calls.. I
have an Audiocodes TDM to IP gateway.. so my PBX is basically sending a call
an auto attendant that they reach that gives them a
personal auto attendant of numbers they can reach - managers, HR, etc.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of David Sharafy
Sent: Monday, November 22, 2010 4:14 PM
To: sipx-users
Hi,
I have a setup that requires users on Sipx to receive calls only and not be
able to call each other..
how can I disable a user from calling another extension?
Thanks
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive:
The documentation I have at hand states that SipX does not support any
CTI features, does this still apply?
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Am 16.11.2010 13:29, schrieb Tony Graziano:
Thats a broad term. Can you explain what you are looking for? 3PCC?
screen pops? DB queries and call routing?
On Tue, Nov 16, 2010 at 7:24 AM, David Becker
david.bec...@itison-ikt.de mailto:david.bec...@itison-ikt.de wrote:
The documentation I
Am 09.11.2010 09:39, schrieb David Becker:
Am 09.11.2010 08:50, schrieb Worley, Dale R (Dale):
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of David Becker
[david.bec...@itison-ikt.de]
Hm, closer
Hm, closer inspection reveals that the Siemens phone doesn't send an
algorithm=md5 parameter. Also the contact address is doubled, is that
problematic?
Am 20.10.2010 10:53, schrieb David Becker:
I found this in the sipstatus.log:
2010-10-20T08:46:36.635044Z:756:AUTH:DEBUG:test.voip.ikt
I had the same problem yesterday, call was dropping after 32 seconds.. my setup
was different though... my issue was the codecs... I was using Alaw instead of
Ulaw... also System -- Internet Calling -- Intranet Subnet make sure that it
is correct...
hope this helps
-Original
Can you update the ede scripts to use these new addresses?
Am 27.10.2010 06:06, schrieb Douglas Hubler:
I apologize for moving builds around as much as i am, i'm trying to
preserve migration paths where i can and create a system that usable
going forward. On that note, i created a couple pages
Can someone please help me with this?
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of David Sharafy
[da...@ccds.ca]
Sent: Friday, October 22, 2010 1:03 PM
To: sipx-users@list.sipfoundry.org
Subject
are getting to sipx, and view the log in the mediant to see
what information it is catching about the call.
On Sat, Oct 23, 2010 at 11:34 AM, David Sharafy
da...@ccds.camailto:da...@ccds.ca wrote:
Can someone please help me with this?
From:
sipx-users-boun
Hi,
I have an Audiocodes M 1000, firmware version 5.2 .. I have been using Asterisk
before and new to Sipx..
Audiocodes has a T1 that is patched to an internal PBX (using a T1 cross over
cable), the pbx is Dialogic based. I'm sending a call from Dialogic to
Audiocodes and then converting it
, David Sharafy
da...@ccds.camailto:da...@ccds.ca wrote:
I thought that it might work with IP.. after configuring a local DNS server, I
now get unauthorized user which is a good sign.. Thanks a lot
From:
sipx-users-boun...@list.sipfoundry.orgmailto:sipx
I found this in the sipstatus.log:
2010-10-20T08:46:36.635044Z:756:AUTH:DEBUG:test.voip.ikt-bs.de:SubscribeServerThread:B6D61B90:SipStatus:SubscribeServerThread::isAuthenticated()
- No Credentials for
mailboxUrl=\sip:2...@test.voip.ikt-bs.de:5060;transport=udp\,
Hi All,
I have installed sipX today and all went well with the installation.. for hours
and I have been trying to register a softphone and just can't make it to work.
I have used Xlite, Zoiper and even Cisco 7940 phones...
the registration is just timing out... I installed wireshark on the
access to domain
registration information that way – srv records, sips, etc.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of David Sharafy
Sent: Wednesday, October 20, 2010 4:53 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users
Can I piggyback onto this thread? I've got the same problem with an
OpenStage phone, the phone sends a SUBSCRIBE, gets an UNAUTHORIZED back
and then sends another SUBSCRIBE with credentials attached which also
get an UNAUTHORIZED response. I've attached a capture file of that
transaction. If
contact address twice and
includes the display name. Removing the display name makes it send the
extension number instead.
It could also be that it sends the credentials incorrectly in some way
though it can register properly.
Am 19.10.2010 10:06, schrieb Nikolay Kondratyev:
David,
i took a look
-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] *On Behalf Of
*David Becker
*Sent:* Tuesday, October 19, 2010 3:51 PM
*To:* sipx-users@list.sipfoundry.org
*Subject:* Re: [sipx-users] Help with SUBSCRIBE for MWI
That is not under my control
is it me or really pages/links are broken on http://www.sipxecs.org/
when I go to Download Apache 2 Test Page powered by CentOS comes up
when I go to installation and then download stable version I get: The
requested URL /pub/sipXecs/ISO was not found on this server.
anyway, Downloads works ok
got it..thanks
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Monday, October 18, 2010 6:19 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] broken pages/links
Sipxecs.org
Hi,
I was actually searching the web for some stuff in regards to Audiocodes and I
cam across this site...
Can I use SipXecs as a SIP registrar? if yes, will it work if it's behind a NAT
and the client is also behind a NAT?
I actually have an Audiocodes TDM to SIP and I'm currently using
to a firewall that
will do this:
See:
http://blog.myitdepartment.net/?p=37
On Sat, Oct 16, 2010 at 10:34 AM, David Sharafy
da...@ccds.camailto:da...@ccds.ca wrote:
Hi,
I was actually searching the web for some stuff in regards to Audiocodes and I
cam across this site...
Can I use SipXecs
, not the local one.
Am 30.08.2010 11:58, schrieb David Becker:
Ran into the next roadblock...
On the development system attempts by the phones to register with
sipXecs are met with 100 Trying responses for a moment before
following with 408 Request timeout. The sipregistrar.log is attached.
All
I'm trying to set up SipXecs on a system for development purposes but
it's somehow misconfigured. I'm using DHCP and DNS on the same machine
but only configured them after installing the EDE and everything.
Running sipxecs-setup-system doesn't fix it, immediately after the
statuses will show
We managed to get this resolved locally, it was a misconfiguration of
the Linux part of the system.
Am 30.08.2010 09:48, schrieb David Becker:
I'm trying to set up SipXecs on a system for development purposes but
it's somehow misconfigured. I'm using DHCP and DNS on the same machine
Ran into the next roadblock...
On the development system attempts by the phones to register with
sipXecs are met with 100 Trying responses for a moment before following
with 408 Request timeout. The sipregistrar.log is attached. All services
are green in the web interface except Media
On Mon, Jul 12, 2010 at 1:18 AM, Graeme Allen
gal...@mytelecom.com.au wrote:
Has the Grandstream GXV3140 been added to the
configuration/provisioning interface of SipX, if not, are
there plans to add it?
As an initial test, have you tried configuring it as a GXV3000?
Can you (or
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To: 4c34fe9b.9020...@gmail.com
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 48871
Message-ID: bee7.4c361...@forum.sipfoundry.org
Yes sipX 4.0.4
And the firmware in my audiocodes is
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To: bac9.4c17f...@forum.sipfoundry.org
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 48831
Message-ID: bebf.4c34f...@forum.sipfoundry.org
I just found your posts, after doing a search.
-Original Message-
From: Staffan Kerker [mailto:ietf-li...@kerker.se]
Sent: Wednesday, June 23, 2010 7:24 AM
To: Saint, David (David)
Cc: sipx-users
Subject: Re: [sipx-users] Sipxconfig on seperate host?
On 18 jun 2010, at 15.07, Saint, David (David) wrote:
I would have
it each time. Can anyone confirm if this
is a problem?
On Mon, Jun 21, 2010 at 3:20 PM, David Minor davemi...@gmail.com wrote:
I figured out how to generate the digest authentication response, and
it turns out the 600v3 is using a blank password. Since the 600v3's
web interface doesn't display
/gethelp/
Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.
--
_
David Minor
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
to domain.
On Mon, Jun 21, 2010 at 2:17 PM, David Minor davemi...@gmail.com wrote:
The phone has been set to the user's password. I set it again to make
sure, but no change.
Is there nothing else it could be?
On Mon, Jun 21, 2010 at 10:24 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
WORLEY, Dale R (Dale)
Sent: Thursday, June 17, 2010 4:24 PM
To: Staffan Kerker; sipx-users
Subject: Re: [sipx-users] Sipxconfig on seperate host?
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 47750
Message-ID: ba86.4c16b...@forum.sipfoundry.org
Hi Im having problems with a couple of polycom soundpoint
500 phones that work great apart of
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 47752
Message-ID: ba88.4c16b...@forum.sipfoundry.org
Configuring the audiocodes gateway seems pretty easy, pretty
much plug and play. I had SipXecs
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Josh Patten
Sent: Thursday, June 10, 2010 7:00 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Audiocodes gateways disable voice
On 05/15/2010 11:54 AM, Josh Patten wrote:
When using SLA/BLA with an Audiocodes Mediant 1000 I noticed that when placing
calls on hold and retrieving them on different phones the caller ID changes
from whatever came over the PRI to the following format:
121 where 1 is the trunk
-Original Message-
From: Mossman, Paul (Paul)
Sent: Tuesday, May 18, 2010 10:19 PM
To: Saint, David (David); sipx-users@list.sipfoundry.org
Subject: RE: More than one VM dial plan rule? (XX-7822)
Dave wrote:
I use two Voicemail dial plan rules when creating a private network
I'm getting ready to write a proposal to replace our existing phone systems
with a sipxecs implementation.
I'm envisioning purchasing 2 T1 gateways - 1 for bringing dial tone in - the
other for linking to our old phone system (during the transition period) - then
re-using the 2nd one as we
.
On Wed, Apr 28, 2010 at 6:50 PM, Worley, Dale R (Dale)
dwor...@avaya.com wrote:
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of David Minor
[davemi...@gmail.com]
If I look at the flow in the SIP
IP PBX -- http://www.sipfoundry.org/
--
_
David Minor
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
packets as above.
So does anyone know if this is normal SIP traffic? Any idea why these
packets are being resent?
--
_
David Minor
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx
Hi Dave,
I have managed to get a wireshark trace of the divert to
voicemail. The key bit of the decode are:
From: 6667912
sip:mailto:6667...@10.203.105.50;tag=d19c5205-82bd-44fc-88c4
-bf5d3c52feb5-37461274
Diversion: 6667912
Thanks Dave,
I didnt see the mailto: bit in the wireshark trace, that may
have been added by one of our browsers. Should the
improvement be able to pick up the divert number even if
there is no dial tag? i.e Cisco seems to format the diversion
hear with the directory number in
Hi All, I am still trying to find answers for these issues.
Update:
On point 3 - I came across a setting on CUCM which enables
the sending of RDNIS across the SIP trunk. The impact of this
now is when a call is diverted to the voicemail pilot number
the voicemail system provides
Hi Dave,
Thanks for your comments, the way you describe seems to be
the method most voicemail platforms work, but either I have
set something up wrong or there may be a bug in the platform.
So, I have the following:
SipXecs - 4.2.0
CUCM 6.1.3
The CUCM has a SIP trunk
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Abdul Mayat
Sent: Tuesday, April 20, 2010 6:12 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] MWI to an external system - is this possible?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Abdul Mayat
Sent: Tuesday, April 20, 2010 11:57 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] MWI to an external system - is this
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Picher, Michael
Sent: Tuesday, April 20, 2010 3:30 PM
To: Josh Patten; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] MWI to an external system
to the Aruba Networks controllers.
Note: wireless controllers are expensive.
My $.02
-Jim
--
_
David Minor
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 42831
Message-ID: a74f.4b8f7...@forum.sipfoundry.org
Hi
This is my first try at setting up a SipXecs pbx. Its for a
small office, only 8 phones(all
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To: a7af.4b90a...@forum.sipfoundry.org
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 42928
Message-ID: a7b0.4b90b...@forum.sipfoundry.org
Ok I was too quick to post.
A google search
Hi,
I was having a discussion with the sipXecs configuration team on the
configuration of SBC/ITSP trunks. In an effort to simplify this further
there was a suggestion for the page:
Gateways / Add new gateway / SIP trunk
We would like to make the SBC Route into a simple drop-down list, e.g.
Using 4.0.4, is there any way to change the forward no answer
duration on an extension?
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p:
I am now able to use the Aastra 9133i, I can dial out and I can
receive calls, I can also receive calls from all extensions, but I can
not dial extensions.
I use a 3 digit extension, If I dial 8 + the extension I go
right into voice mail.
Dan
You can setup (as a user) distribution lists.
Their is a tracker item for what you are describing and is
being worked on.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
...
Hi there:
On our old PBX system, we were able to send a
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Lawrence, Scott AVAYA (BL60:9D30)
Sent: Tuesday, January 05, 2010 8:11 AM
To: Josh Patten
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Wiki
I'm also after the capability to have have conference calls
recorded, and then emailed to the owner of the conference.
Any constructive suggestions, or pointers to documentation
that I've missed would be appreciated.
Cheers
Arne
The conference recording feature is currently under
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
James Johnson
Sent: Friday, October 02, 2009 2:31 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Auto Attendant
How do I set up so when you
-Original Message-
From: Jonathan Petersen [mailto:jonathan.peter...@ontraonline.com]
Sent: Tuesday, August 04, 2009 4:06 PM
To: Saint, David (CAR:9D60); sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] User expectation for call forward
functionality
Huh, very
Hi,
A question came up during our code review as to what functionality the
sipx
users would expect in the following call forward scenerio:
Configuration:
User 201
Extension 201 will ring first for 5 seconds
If no response forward to 202 ring for 30 seconds
If none of the above
and make some test calls.
Cheers,
David
- Original Message -
From: M. Ranganathan mra...@gmail.com
To: David Hobley david.hob...@mionegroup.com
Cc: sipx-users@list.sipfoundry.org
Sent: Saturday, 30 May, 2009 2:21:08 AM GMT +10:00 Canberra / Melbourne /
Sydney
Subject: Re: [sipx-users
Hello David,
after enabling the option I'm getting this message for all
test starting from 'DHCP test':
DHCP Test 5/19/09 9:20 PM Warning
Unknown test result: 1
* Hide details
Starting DHCP server test.
Sending DHCPDISCOVER request.
DHCPOFFER responce
Note that the DHCP option 120 on SIPx is intended for use by the Bria
Professional and Nortel SMC3456 softphones,
the content should be a text string that looks like this (replacing
example.com with your server name):
example.com:12000/cmcprov/login
or
Hallo,
after running all Configuration Diagnostic Tests I got the
error DHCP (Option 120) Test No SIP servers supplied.
After enabling the option in windows 2008 DHCP-Server I got
the error Unknown test result: 1.
The standalone programm preflight never stops DHCP Server Test.
connecting with Asterix at the ITSP
end?
Cheers,
David
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
Restarting did not help here.
I made a second HA installation on another System. Same issue.
Maybe a general Problem with the 4.0 release?
René
2009/4/29 Kevin Thorley kevin.thor...@nortel.com
On Wed, 2009-04-29 at 11:25 +0200, Rene Pankratz wrote:
I recently
If the Mediatrix were able to add the leading digits to CLID it would
have to be done based on the CLID's NPI/TON.
For example in the UK if the CLID is E.164/National it would need to add
a leading 0, if the CLID is E.164/International it would need to add a
leading 00, if the CLID is
87 matches
Mail list logo
