We setup most of our deployments with an SSLVPN on port 443. That is allowed
out of all WIFI spots and the vpn removes any special "remote-worker" config.
-M
>>> On 9/30/2009 at 6:46 PM, in message <200993017462.149...@leena>,
>>> "li...@grounded.net" wrote:
After testing for a solid wee
After testing for a solid week, we're floored to come to a realization... MOST
hot spot providers are either blocking voip or don't have ALG or other problems
which prevent voip from working properly. The main culprit was one way audio.
So, after visiting at least a dozen sites and only one wor
I think freeswitch plays media files. For acd, it interacts (plays
prompts I think), so yes, its involved.
On 9/30/09, James R wrote:
> On Wed, Sep 30, 2009 at 1:40 PM, Dale Worley wrote:
>
>>
>> They result from a process that crashed, as you know. Sometimes
>> processes crash when sipXecs is
On Wed, 2009-09-30 at 22:22 +0200, Staffan Kerker wrote:
> That's the problem, the log files are gone, but I will try to redo the
> call tomorrow
> and attach the SipX-log.
>
> I have a Wireshark trace, but since it's not showing the interal call
> flow
> of SipX it might not be that interesti
On Wed, Sep 30, 2009 at 1:40 PM, Dale Worley wrote:
>
> They result from a process that crashed, as you know. Sometimes
> processes crash when sipXecs is shut down, which isn't so important.
> But all other crashes are bugs that we'd like to see eliminated. So if
> these don't correlate with re
On Tue, Sep 29, 2009 at 11:49 PM, Tony Graziano <
tgrazi...@myitdepartment.net> wrote:
> Ah. details. Thanks. Running ACD server? Are you making changes to the ACD
> system? Is the agent on the same local segment as the acd server?
>
Yes we running the ACD server... we are only using sipX for it
That's the problem, the log files are gone, but I will try to redo the
call tomorrow
and attach the SipX-log.
I have a Wireshark trace, but since it's not showing the interal call
flow
of SipX it might not be that interesting. It only shows the received
"420" '
from the Tandberg UA and then
Hi, The system is getting a lot of core dump files, that are filling the
hard drive constantly. I'm thinking that these
core files are relationated with the SIPConf service that appear in FAILED
state.
All the files are located in the next route:
*/usr/local/conferencing_sipX-3.3.186/sipconf*
*
Actually, if a wiki page for the config were to be created, that would help
a lot of folks.
On Wed, Sep 30, 2009 at 4:14 PM, Bernardo Ortega wrote:
> Hi Todd, our problem with PhoneEasy is when received a call, I click on
> answer button and only show me a messages "please wait..." and nothing
>
On Wed, 2009-09-30 at 21:59 +0200, Staffan Kerker wrote:
> I'm having an issue with Tandberg terminals, but I have to agree with
> them that it seems to be a SipX issue.
>
> 1. SipX receives initial INVITE with a Require-header for a Tandberg
> proprietary feature.
> 2. SipX forwards the INVIT
Hi Todd, our problem with PhoneEasy is when received a call, I click on answer
button and only show me a messages "please wait..." and nothing happend. If you
can show your configuration to compare with ours would be helpful.
Thanks,
- "Todd Hodgen" wrote:
|
|
I can confirm that the
Evan,
VG-200 is too old, if you Google VG-200 & SIP, you get this article,
http://www.cisco.com/en/US/docs/ios/12_2t/12_2t13/feature/guide/ftspdrcr
.html
which states: Note The Cisco VG200 does not support the session
initiation protocol (SIP).
Get yourself a 2811/2821 series router or a new
I can confirm that the latest version of PhoneEase does in fact work with
sipXecs. If your’s is not working, I would look at another issue in your
setup. Finding another console may not be the answer, but a trail leading
right back where you started.
From: sipx-users-boun...@list.sipfoundr
Hi
I'm having an issue with Tandberg terminals, but I have to agree with
them that it seems to be a SipX issue.
1. SipX receives initial INVITE with a Require-header for a Tandberg
proprietary feature.
2. SipX forwards the INVITE to the registered UA
3. The UA responds correctly with a "420 B
Hi,
Somebody in the list has know some Operator Panel that works with SIPxecs?. In
this moment we try:
1. PhoneEasy IP Console and don't answer the call.
2. Voice Operator Panerl and don't register in SIPxecs, generate an error
unable to register .
If anybody know other soft, pleas
Has anyone heard or ever used the Cisco VG-200 as a PRI Gateway with
SipX?? I currently have the VG-200 connected to my Callmanager, but I
am looking to junk the CCM for the SipX server and would like to reuse
the VG-200. Any suggestions?
Regards,
Evan Gottlieb
Network Engineer
Johnston C
Did you make sure your proxy and registrar is running in sipxconfig, does
anything user the server show failed? is the phone using the same DNS server
as the sipxecs system? Can you register a softphone like xlite to it?
On Wed, Sep 30, 2009 at 3:17 PM, Derek Bartolo <
dbart...@comsourcetechnologi
I am getting the following error when I try to click on users or phones:
An internal error has occurred. Click here to continue.
I am running Sipx 4.0.2. On a Centos 5.3. As well as root.
..
James
___
sipx-users mailing list sipx-users@list.sipfoun
Personally, I find ftp to be less problematic. Did you add the user account
(I.e. Extension 200) to the phone and then send the profile first?
When it boots it will detect your server has updated bootrom and/or firmware
and load that first if available. Then it grabs the generated profile then
reg
Did you generate and send the profile to the phone with the line added
first? Are you configuring them to pick up the profile via tftp or ftp? The
first registration will require a manual reboot of the phone 9or until it is
registered) before it will pickup the reboot requests.
url call disabled is
Hey Guys I just loaded up sipx 4.0.2 on a box and tried to get 2 polycom
IP330s running but after the phones come up with the proper extensions I
get "url call is disabled" and none of the phones are showing up
registered on the sipx box.
Anyways maybe I am missing/forgetting to do something b
On Tue, 2009-09-29 at 17:17 -0400, Bernardo Ortega wrote:
> Thanks for the answer, the only problem now is when we try to answer a
> call in IP Console, If I click on answer button only appear "Please
> wait..." messages and do not answer the call.
You will have to trace a failed call and examine
On Tue, 2009-09-29 at 23:06 -0400, James R wrote:
> I've been spending most of my days looking through the log files
> in /var/log/sipxpbx and usually I end up having more questions than I
> get answers. Are there any developer guides, documentation, details
> etc on the log files?
Not much. As
Huw W. Jones wrote:
> Some more things that might help ...
>
> I notice from the Audiocodes interface that the four FXS ports are 'not
> registered' - presumably with SipX. However the gateway settings in SipX
> don't give me the option of registering the ports. Have I missed
> something here?
>
I think he's referring to the line (DS3 / T3) coming into the gateway,
rather than a gateway manufacturer. That said, the Cisco 3845 appears
to support operating at T3 rates and should be able to act as a
gateway. However, I have not used this router, and I believe other
users on the lis
James, I would like to help you but in order to do so, I'll need more
precise info. Please see my comments inline.
> Sent: Tuesday, September 29, 2009 9:32 PM
>
> Our agents pass through a box that has 2 physical Ethernet
> interfaces before getting to sipX.
> 1 DMZ and 1 inside
Wouldn't th
I assume the analog lines are analog phones (especially since we are
talking about FXS ports)?
Usually when we talk lines, we are talking about circuits from a telecom
provider. These go into FXO ports and don't need to register.
I'm not familiar with the AC 1000 enough to tell you how to do thi
My opinion is that it is easier to manage mediant manually, using it's own
web interface. Just imho...
Nikolay.
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
> boun...@list.sipfoundry.org] On Behalf Of Huw W. Jones
> Sent: Wednesday, September 30,
Some more things that might help ...
I notice from the Audiocodes interface that the four FXS ports are 'not
registered' - presumably with SipX. However the gateway settings in SipX don't
give me the option of registering the ports. Have I missed something here?
Huw
- Original Message ---
Hi,
Over the past ten days or so we've been installing a new PBX system in the
college where I work. We're using SipX along with an Audiocodes Mediant 1000
gateway. So far everything is running pretty smoothly - except for integration
of old analog lines into the system. Our Mediant 1000 has an
I would take silence as a no. I've never heard of them.
I usually recommend Patton.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Gerald
Drouillard
Sent: Tuesday, September 29, 2009 9:17 AM
To: sipx-user
Absolutely.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Huw W.
Jones
Sent: Wednesday, September 30, 2009 4:35 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] List etiquette
Hi folks
Is this a sui
Hi folks
Is this a suitable list on which to ask the occasional 'support' type question?
TIA
Huw
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