Thank you for your reply Mr. Picher,
Well I'm certainly not going to discount that as a possibility.
Before experimenting with sipXecs we were using trixbox CE which VoicePulse
supports directly. Basically, you just download a module written by
VoicePulse, install it on your trixbox serve
Ok, my bad sorry for the mis-understanding on my half..
I have the Australian did 0382569691 active with ip forwarding to my sipx
ip address in my itsp web-admin.
Now i have a user extension with the ID '0382569691' on my sipx server.
I have a IP phone registered to that user extension on the sipx
On Mon, 2009-06-08 at 13:53 +1000, dned...@flexinet.com.au wrote:
> Yes, basically my ITSP is sending all calls to my did's directly to my sipx
> ip address..
> So when a call comes in, sipx recognizes the inbound call but blocks it
> because the ip address(es) from my ITSP do not have permissions
Yes, basically my ITSP is sending all calls to my did's directly to my sipx
ip address..
So when a call comes in, sipx recognizes the inbound call but blocks it
because the ip address(es) from my ITSP do not have permissions on my sipx
server. This used to work fine in sipx v3, its only in the lat
On Sun, Jun 7, 2009 at 7:20 AM, Boy Aidil
Sjam wrote:
> How if I want to setup one of the server as an ITSP?
>
> In the "SIP Trunking with sipXecs: Overview and Configuration"
> (http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration),
> only describe how the
On Mon, 2009-06-08 at 13:14 +1000, dned...@flexinet.com.au wrote:
> OK, basically i wanted to know what i can do to give permission to those
> sources/ip's so that calls can come through.
You can't. sipXecs does not support authentication by IP address.
What is it that you're trying to do?
___
Thanks for the answer Scott.
I hope this feature will be supported in the future.
> -Original Message-
> From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
> Sent: Monday, June 08, 2009 10:01 AM
> To: Boy Aidil Sjam
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] In
OK, basically i wanted to know what i can do to give permission to those
sources/ip's so that calls can come through.
On Sun, 07 Jun 2009 22:52:44 -0400, "Scott Lawrence"
wrote:
> On Sun, 2009-06-07 at 12:35 +1000, dned...@flexinet.com.au wrote:
>> Ok, i've uploaded a siptrace file for 0382569691
On Sun, 2009-06-07 at 22:41 +0200, Alberto wrote:
> Hi,
> i was reading an old post where Dale was explaining how to extract a
> single call from logs:
>
> /usr/bin/merge-logs --containing= /var/log/sipxpbx/sip*.log
>
> where is identifying a single call.
>
> Don't know if current CDR is actua
On Sun, 2009-06-07 at 07:45 -0700, hassen yahiaoui wrote:
> PLEASE
> CAN I CAN LOGIN MY SESSION ;I HAVE PB WITH MY SIPFOUNDRY 4.0
[no need to shout]
What is it you want to log in to? and what is your problem?
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On Sun, 2009-06-07 at 18:20 +0700, Boy Aidil Sjam wrote:
> How if I want to setup one of the server as an ITSP?
>
> In the "SIP Trunking with sipXecs: Overview and Configuration"
> (http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration),
> only describe how
On Sun, 2009-06-07 at 12:35 +1000, dned...@flexinet.com.au wrote:
> Ok, i've uploaded a siptrace file for 0382569691...
> it shows the "SIP/2.0 407 Proxy Authentication Required errors" Im
> getting...
These traces are consistent with what you see when calling the
destination requires some permis
On Sun, 2009-06-07 at 12:36 -0400, Matt Keys wrote:
>
> I've got incoming calls to the ACD queue from the MP-114 working but
> the caller ID information isn't being displayed--only the number used
> for the Auto-Dial to call the queue. How do I fix this? Attached is
> the MP-114 configuration INI
Hi Tony,
Thanks for the prompt reply. I have not fully tested out the voice and
video call using Windows Messenger. The problem now is that I can't see
the online presence of the users that I have added. I have read about the
presence server in sipX. Is there any workaround for this? I have
config
Voice pulse probably expects you to send more than seven digits. Your dialplan
needs to prepend 7 digit calls with 757.
-Original Message-
From: "Picher, Michael"
To: Andreas (Around the Clock Information Systems)
To:
Sent: 6/7/2009 8:50:09 PM
Subject: Re: [sipx-users] Is there some
I'm not that familiar with the Cisco phones. Is there a Dial plan in
the phone that tells the phone when to send the dialed #?? It is
something we have to tweak with the Polycoms sometimes.
Mike
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
> bo
Hi,
i was reading an old post where Dale was explaining how to extract a
single call from logs:
/usr/bin/merge-logs --containing= /var/log/sipxpbx/sip*.log
where is identifying a single call.
Don't know if current CDR is actually storing call-id but the idea might
be quite simple. Why don't j
Dear sipXecs Users, Experts and Developers,
Well, after a couple rough weeks of getting my sipXecs (Build
4.0.0-015321) Server configured, I think I've gotten the core essentials
working well enough to do some real testing.
Currently, I am able make and receive calls to/from every
On Sun, 2009-06-07 at 15:32 -0400, Dale Worley wrote:
> On Sun, 2009-06-07 at 15:25 -0400, Matt Keys wrote:
> > > You're sending a REGISTER inward through sipXbridge, which should
> be
> > > OK
> > > in principle. Though the Asterisk is trying to register itself as
> > > "sip:7062914...@pbx.pfnnet
On Sun, 2009-06-07 at 13:47 -0400, Matt Keys wrote:
>
> Regarding my original email here's some more detail. I was reading the
> information about the from header here:
> http://sipx-wiki.calivia.com/index.php/SipX_Phone_Registration_Troubleshooting
> .. it appears to be the same.
>
> REGISTER s
Polycom 330 & 650's. I don't bother with the 320's & 550's.
The 330's can be had for under $115 (US) and the 650's are around $250
(US).
You definitely get what you pay for with phones...
Mike
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
> bo
Regarding my original email here's some more detail. I was reading the
information about the from header here:
http://sipx-wiki.calivia.com/index.php/SipX_Phone_Registration_Troubleshooting
.. it appears to be the same.
REGISTER sip:pbx.pfnnet.net:5080 SIP/2.0
Via: SIP/2.0/UDP
208.71.234.170:
In your opinion which is the best ip phone of the market
?...Considering qualiy/price ...thank you
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Windows Messenger is not exactly what you might be looking for. In sipx version
3.8 you could configure Windows Messenger. Since MS uses TCP for a lot of voice
and messaging, it never really worked right, as you would have to re-add users
every time you started the program (at least that's what
I've got incoming calls to the ACD queue from the MP-114 working but the caller
ID information isn't being displayed--only the number used for the Auto-Dial to
call the queue. How do I fix this? Attached is the MP-114 configuration INI.
Thanks,
Matt
00908F1D34BD.ini
Description: 00908F1D34BD.
PLEASE
CAN I CAN LOGIN MY SESSION ;I HAVE PB WITH MY SIPFOUNDRY 4.0
SLT
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Hi developers and fellow sipX users,
I have installed SIPX on my CentOS 5.2. I successfully configured Windows
Messenger 5.1 to login to the server. However, I do not see any presence of
other users that I have added in my contacts. I'm fairly new to SIP and
SIMPLE. I hope anybody here can help
How if I want to setup one of the server as an ITSP?
In the "SIP Trunking with sipXecs: Overview and Configuration"
(http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration),
only describe how the sipXecs server to interconnect with ITSP.
Thanks,
B. Aidil
Personally, with the price of standard PC type hardware vs. Mac
hardware, why bother? Your PBX should reside on a standalone machine
anyway for the best voice (voicemail / AA) quality.
But, if you're volunteering, go for it!
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sip
I had tried some of their phones a while ago and was not happy with the
quality...
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
> boun...@list.sipfoundry.org] On Behalf Of alessio flaiban
> Sent: Saturday, June 06, 2009 11:21 AM
> To: sipx-users
I think leveraging FreePBX is something that has been talked about for
the Skype connectivity. I think what Tony was referring to was Skype's
allowing SIP traffic into their system. I tried signing up for the Beta
a couple months ago but I was too late.
Mike
> -Original Message-
> From:
Thanks Tony...
Maybe I can garner enough information from the gateway logs to get some
help from Patton.
Mike
> -Original Message-
> From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> Sent: Saturday, June 06, 2009 4:02 PM
> To: Picher, Michael; sipx-users@list.sipfoundry.org
> C
Music on Hold? Not that I've heard.
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
> boun...@list.sipfoundry.org] On Behalf Of alessio flaiban
> Sent: Sunday, June 07, 2009 4:47 AM
> To: sipx-users@list.sipfoundry.org
> Subject: [sipx-users] grands
Does the grandstream's product support the hold down music ??
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