Hi,
I Install the latest development of sipxecs to go test the integration with
xmpp, but I could not find the part where you manage user groups as in xmpp
openfire as such, would appreciate much any kind of commentary on this part.
Thanks,
___
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ry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Bernardo Ortega
| Sent: Wednesday, September 30, 2009 12:52 PM
| To: SIPxecs Support
| Subject: [sipx-users] Soft Operator Panel
Hi,
|
| Somebody in the list has know some Operator Panel that works with SIPxecs?.
I
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
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.
- "Dale Worley" wrote:
| On Tue, 2009-09-29 at 09:47 -0400, Bernardo Orteg
How to configure blf services in sipxecs?, currently we have installed an
Aastra 57i for operators with an M675i Expansion Module but, I don't know how
to configure BLF in SIPxecs and Aastra. We want to configure, because we are
going to test PhoneEasy IP Console from Advantel.
Thanks,
__
Hi,
Where can I get an ISO file from version 3.8.1?
Thanks,
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sip
We have the following installation:
• SIPxecs 4.0.2
• Polycom 320/330
• Audiocodes TP260
The problem comes when I call to autoattendant from external, the call is only
transferred to the operator not to the selected extension but, if the user has
a DID assigned can receive the
Hi Josh,
We are still with the problem of freezing phones, we tried installing a version
of similar in sipxecs everything that we have in production, the only
difference was that we use a switch without PoE and every phone we put your AC
adapter, but still be freezing. We have BootROM 3.1.3C
We have the following platform:
• SIPxecs 4.0.2
• Polycom 320/330
• Audiocodes TP260
The problem comes when I call autoattendant, the call is only transferred to
the operator not to the selected extension but, if the user is assigned a DID
if you can receive the call via the au
Anybody know in what version we can wait the account codes for make
international call or if exist any way for make that?
Thanks,
Bernardo
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Hi Cuneyt sorry for my english (I try to say lost record in wins record ). We
try the firmware 2.2.2 and bootrom 4.1.0 but not work, the polycom freeze on
boot and need back to 3.1.3c and rom 4.1.0.
Bernardo.
- "Cuneyt M" wrote:
| Hi Bernardo and Michael,
|
| We use that model of switc
at’s where he’s
from). He’s using Linksys PoE network switches.
What are you using for network equipment?
Mike
|
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Bernardo Ortega
| Sent: Saturday, September 12, 2009 9:53 AM
p/relnotes/spip_ssip_3_1_3RevC_relnotes.pdf
|
| See page 5.
| -Original Message-
| From: Bernardo Ortega
| To: SIPxecs Support
|
| Sent: 9/11/2009 9:18:32 PM
| Subject: [sipx-users] sipxecs 4.0.2 and polycom 320/330
|
| I have the following scenario:
|
| • Server: SIPxecs 4.0.1
I have the following scenario:
• Server: SIPxecs 4.0.1
• Phone : Polycom 301/320/330 PoE
• Firmware: 3.1.0C
• BootROM: 4.1.0
The problem is this:
• When the phone is time wins record, is seeking a new central register but
fails and freeze, then has to be restarted by
We have installed SIPxecs 4.0.1 and polycom phones are 320/330 with fimware
bootROM 3.2.3 and 4.1.2, I do not understand why the phone when is going to
reregister they stay in the air though they have 3600 seconds which is the
default time.
Thanks,
___
Hi,
We have installed SIPxecs 4.0.1 with AudioCode M1000, the problem is with the
call transfers with autoattendant only work when the user is assigned a DID,
when DID remove the call is not transferred via autoattendant rings until you
leave voice mail.
Thanks,
we made the changes that Tony Graziano said, but unfortunately we do not
have the expected result, we continue with the problem of incoming
calls. If we have an active call and enter a new call, the new call cut
the first call.
Bernardo Ortega wrote:
Right now the setup is in silence, I
quot;devices" and pull up one of the phones. click on
> sound effects. your call waiting should be set to "chord".
>
> What is the current setting?
>
> >>> Bernardo Ortega 07/19/09 12:49 AM >>>
> I have serious problems with the dual line te
I have serious problems with the dual line telephones, if I have an
active call and a new call comes, I cut the incoming call. I have
installed 3.10.2 with Polycom 320/330 Phones with a AUDIOCODES M1000 I
have installed this version and the version 4.0.1 causing similar
problems and I thought back
/mailman/listinfo/sipx-users
> sipXecs IP PBX -- http://www.sipfoundry.org/
--
Bernardo Ortega
Email: jbort...@fschad.com
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We had to install SIPxecs 3.10.2 to solve some compatibility problems,
but after the installation when they transfer a call to a user with an
open line, the new call cuts the existing call. We have reviewed all the
parameters in the phone and look good, the only thing is that they only
have one as
Hi,
We installed sipxecs 4.0.1 (stable) and services freeswitch and
sipxproxy take 99% of the processor which is causing us problems with
internal and external calls, so far I have not managed to find the
reason of this problem. Any help with this problem is welcome.
__
Hi,
Some many day, we migrate to SIPxecs 4.0.1 from 3.10.3 after this the AA
left to work, in this moment when dial extension 100 and then try to
dial any extension only forward calls to the operator, and I have
reviewed all the settings are as they were in the version 3.10.3 and I
do not generate
Hi,
Why when I try to active phonelog, all phone in the network close the
registration with the server?
Thanks,
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Hi,
I received this messages some many times in sipXproxy log:
"2009-06-28T01:57:19.401569Z":872519:SIP:ERR:pbx2.fschad.com:SipUserAgent-2:B6D67B90:SipXProxy:"SipUserAgent::handleMessage
SIP message timeout expired with no matching transaction"
I think that is the initial of all my problems. If
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