I'll vote for that :-)
> -Original Message-
> From: Todd Hodgen [mailto:thod...@verizon.net]
> Sent: Tuesday, November 10, 2009 2:46 PM
> To: 'Boy Aidil Sjam'; sipx-users@list.sipfoundry.org
> Subject: RE: [sipx-users] Limiting User Call between remote site
Thanks Todd,
I think that's a good solution for this time being.
I hope there will be a better solution in the sipx system in future.
> -Original Message-
> From: Todd Hodgen [mailto:thod...@verizon.net]
> Sent: Monday, November 09, 2009 11:28 PM
> To: 'Boy Aidil
,
B. Aidil
> -Original Message-
> From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
> Sent: Monday, November 09, 2009 6:51 PM
> To: Boy Aidil Sjam
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] Limiting User Call between remote site
>
> On Mon, 20
Hi All,
A while ago , I once read someone asking "whether there is a method of sipx
to restrict calls between the gateway", and I know there will new features
to restrict dialing between sipXbridge on sipXecs ver 4.2
(http://track.sipfoundry.org/browse/XX-6276). And that's a good news.
But I h
thanks guys for all the respons. I think that's make sense.
I will try using HA solution.
Thanks,
B. Aidil
-
Original Message:
From: Robert Joly
To: Boy Aidil Sjam , sipx-users@list.sipfoundry.org
Cc:
Date: Wednesday, 04 November 2009 21:35
Subject: RE: [sipx-
Hi,
I like to know, is there possible for sipX to have multiple NAT connection?
Let say, I have 2 connection to the internet. I used multi-homing connection
using BGP to the internet.
Both link terminated with NAT from firewall, and I like to utilize both
link to support voip for remote worker.
I am having the same problem some time ago. The difference is I use another
server sipXecs as ITSP.
Try to use DNS domain name in the ITSP domain name, and use ip address or
full hostname (with domain) in ITSP server address.
I hope that could help
From: Emery ville [mailto:emeryvill
-Original Message-
From: Boy Aidil Sjam [mailto:aidils...@prawedanet.co.id]
Sent: Monday, June 29, 2009 12:45 PM
To: 'jbort...@fschad.com'
Subject: RE: [sipx-users] sipXecs CD 4.0.1 fresh install http problem
I Looked in the ssl folder and found only ssl.crt and ssl.key
Gen
Sorry, what I mean is there's no sipxconfig.log file in the
/var/log/sipxpbx/ folder
> -Original Message-
> From: Boy Aidil Sjam [mailto:aidils...@prawedanet.co.id]
> Sent: Sunday, June 28, 2009 9:53 AM
> To: 'Scott Lawrence'; 'tgrazi...@myitdepartment.net
T21:02:20.239184Z":23946:SUPERVISOR:ERR:voip2.net.lab:SipxProcess
-19:B67F6B90:Supervisor:"SipxProcess[ConfigServer]::commandOutput ' at
org.sipfoundry.sipxconfig.common.SystemTaskRunner.runMain(Unknown Source)'"
"2009-06-27T21:02:20.239210Z":23947:SUPERVISOR:ERR:v
,
"SipXbridge"=>"Disabled",
"MediaServer"=>"Disabled",
"sipXivr"=>"Disabled",
"PageServer"=>"Disabled",
"PresenceServer"=>"Disabled",
"ResourceListServer"=>"Disabled
ing localhost address configured:success
Checking localhost name is not shared:success
Checking /tmp directory has correct permissions:success
Starting sipXpbx:
Starting sipxsupervisor: success
Starting httpd: success
Run sipxproc again, but still got the same messages like above
> --
Hostname or ip address got the same result
> -Original Message-
> From: Matt Keys [mailto:mk...@parkersystems.net]
> Sent: Saturday, June 27, 2009 4:24 PM
> To: Boy Aidil Sjam
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] sipXecs CD 4.0.1 fresh in
Hi,
I just fresh installed sipXecs CD 4.0.1 (sipfoundry-4.0.1-015823-i386.iso)
All processed running smooth. But I can't access web UI after installation.
Status from firefox "Failed to Connect"
Tried installed in the different server, got the same messages.
Can someone help me ?
Thanks,
B
4:14 PM
To: Boy Aidil Sjam
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Limiting voice call session
Hi,
you currently focused this issue better than I did. I thought about the
bandwidth management as sipxbridge problem, but definitely I realize is more
general than that.
There is a
Hi All,
Let say, I have two sipXecs in different domain or one sipXecs server
connected to other SIP server (ITSP) or gateway. The connection between both
location is separated with WAN connection with limited bandwidth to share
between the voice call and others data application (256kbps). I had
Ronny,
Thanks for the info.
I can managed SPA941 phone from sipXecs server now
B. Aidil
From: Ronny Tjoa [mailto:pingk...@yahoo.com]
Sent: Thursday, June 25, 2009 2:30 AM
To: Boy Aidil Sjam
Cc: sipx-users@list.sipfoundry.org
Subject: Re: Linksys SPA941 IP Phone - Provisioning Issues
Yup,
I double checked that
> -Original Message-
> From: Keith Gearty [mailto:ke...@glensound.co.uk]
> Sent: Wednesday, June 24, 2009 9:14 PM
> To: Boy Aidil Sjam
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] Linksys SPA941 IP Phone - Provisioning Is
Oops, I mean, I found spa000E08D09436.cfg at the tftproot folder. Sorry for the
typo
-
Original Message:
From: Keith Gearty
To: Boy Aidil Sjam
Cc: sipx-users@list.sipfoundry.org
Date: Wednesday, 24 June 2009 19:25
Subject: Re: [sipx-users] Linksys SPA941 IP Phone
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, the IP Phone was discovered with the
correct mac address, IP address, and Vendor, but there's no single model I
can choose.
Second, I manually a
Another dumb question.
Is it safe if I update the OS too?
Right now I use sipXecs 4.0 CD
> -Original Message-
> From: an...@iguanait.com [mailto:an...@iguanait.com]
> Sent: Wednesday, June 24, 2009 2:58 PM
> To: sipx-users@list.sipfoundry.org
> Cc: Charles
> Subject: [sipx-users] updati
@nortel.com]
> Sent: Friday, June 12, 2009 9:25 PM
> To: Boy Aidil Sjam
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] Interconnection 2 sipXecs - solved
>
> On Fri, 2009-06-12 at 10:03 -0400, Scott Lawrence wrote:
> > On Fri, 2009-06-12 at 17:22 +0700, Boy A
> -Original Message-
> From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
> Sent: Friday, June 12, 2009 9:25 PM
> To: Boy Aidil Sjam
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] Interconnection 2 sipXecs - solved
>
> On Fri, 2009-06-
pXecs very dependent with DNS SRV.
Thanks for the help Scott.
Next step, testing between 2 server with both server behind NAT.
Wish me luck ;-)
Regards,
B. Aidil
> -Original Message-
> From: Boy Aidil Sjam [mailto:aidils...@prawedanet.co.id]
> Sent: Friday, June 12, 2009 12:13 P
> -Original Message-
> From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
> Sent: Friday, June 12, 2009 1:43 AM
> To: Boy Aidil Sjam
> Cc: sipx-users@list.sipfoundry.org
> Subject: RE: [sipx-users] Interconnection 2 sipXecs
>
> On Thu, 2009-06-11 at 11:
l directly terminated.
Is that the way how the sipXecs works or is there something wrong with my
configuration?
Thanks
> -Original Message-
> From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
> Sent: Thursday, June 04, 2009 8:29 PM
> To: Boy Aidil Sjam
> Cc: sipx-
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
> Subjec
,
B. Aidil
> -Original Message-
> From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
> Sent: Thursday, June 04, 2009 8:29 PM
> To: Boy Aidil Sjam
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] Interconnection 2 sipXecs
>
> On Thu, 2009-06-04 at
nfiguration),
only describe how the sipXecs server to interconnect with ITSP.
Thanks,
B. Aidil
> -Original Message-
> From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
> Sent: Thursday, June 04, 2009 8:29 PM
> To: Boy Aidil Sjam
> Cc: sipx-users@list.sipfoun
Hi All,
I have several questions about interconnection between 2 sipXecs server
Here are the scenarios:
1. sipX server A configured for the HQ in the Asia and providing VoIP
services for the local users.
2. sipX server B configured for the Branch office in the Africa and
providin
> -Original Message-
> From: Robert Joly [mailto:rj...@nortel.com]
> Sent: Monday, June 01, 2009 8:21 PM
> To: Boy Aidil Sjam; sipx-users@list.sipfoundry.org
> Subject: RE: [sipx-users] sipXecs 4.0 Registration status for Remote
> Workers
>
> > Hi All,
>
t would not be
> behind NAT. In that instance the registration would show the private IP
> address with "x-sipX-privcontact=xxx.xxx.xxx.xxx (where xxx.xxx.xxx.xxx
> is the address the phone has).
>
>
> >>> "Boy Aidil Sjam" 05/31/09 10:47 PM >>>
&g
e 01, 2009 10:46 AM
> To: Boy Aidil Sjam
> Subject: RE: [sipx-users] sipXecs 4.0 Registration status for Remote
> Workers
>
> Ekiga is a good one: http://www.ekiga.org
>
> There's a win32 version out there too.
>
> On Mon, 2009-06-01 at 10:43 +0700, Boy Aid
Hi All,
I have a question about registration status in diagnostic tab for remote
worker.
1. If a remote user registered to the server with
"3...@114.58.107.77:26116;rinstance=61e4a43ba568d13d;x-sipX-nonat" status,
the remote user able to call the others extension and vice versa.
2.
5060
Thanks all,
B. Aidil Sjam
> -Original Message-
> From: Mark Gertsvolf [mailto:ma...@nortel.com]
> Sent: Monday, May 25, 2009 5:34 AM
> To: Boy Aidil Sjam; Robert Joly; tgrazi...@myitdepartment.net
> Subject: RE: [sipx-users] Media Relay Service Failed
>
> I am
session context
associated with handle '1920-52'"
> -Original Message-----
> From: Mark Gertsvolf [mailto:ma...@nortel.com]
> Sent: Friday, May 15, 2009 11:25 PM
> To: Boy Aidil Sjam; sipx-users@list.sipfoundry.org
> Cc: Robert Joly
> Subject: RE: [sipx-users
need.
>
> Follow Robert's advise on the remote workers cheat sheet, it will make
> a difference.
>
> >>> "Boy Aidil Sjam" 05/15/09 10:39 AM >>>
> Yes, I used static public IP address
> No, I don't use sip trunking (for current time, maybe in
@nortel.com]
> Sent: Friday, May 15, 2009 9:17 PM
> To: Boy Aidil Sjam; sipx-users@list.sipfoundry.org
> Subject: RE: [sipx-users] Media Relay Service Failed
>
> > Right now, I'm still open all UDP & TCP port for the SIPX server.
> > I used static ip address for the
ports and make this work.
>
> You need ports 5060 nad 3-31000 on your WAN port forwarded to your
> private ip address of your sipx system.
>
> You might need others too depending on whether or not you are using sip
> trunking.
>
> Do you have a static public IP addres
Right now, I'm still open all UDP & TCP port for the SIPX server.
I used static ip address for the SIPX server NAT
> -Original Message-
> From: Robert Joly [mailto:rj...@nortel.com]
> Sent: Friday, May 15, 2009 8:25 PM
> To: Boy Aidil Sjam; sipx-users@list.sipfoun
I saw the different RTP packet stream with this configuration
Remote Worker (public address) < RTP ---> Internal user (private
address)
Can someone help me with this problem?
B.Aidil
> -Original Message-
> From: Boy Aidil Sjam [mailto:aidils...@prawedanet.co.id]
&g
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Monday, May 04, 2009 12:05 AM
To: Boy Aidil Sjam
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Media Relay Service Failed
On Sun, May 3, 2009 at 6:55 AM, Boy Aidil Sjam
wrote:
> Hi,
>
> Aft
Hi,
After failed to upgrade sipxecs 4.0 from 3.10.3, I was able to fresh
installed 4.0 from stable ISO.
All the configuration I set already running, including supporting for remote
worker. Its great
But after a while I saw the Media Relay Service status Failed, then I click
show details à see
).
-Original Message-
From: "Boy Aidil Sjam"
To: Tony Graziano
To:
To:
Sent: 5/2/2009 9:46:39 AM
Subject: RE: [sipx-users] Unable to get 4.0 started after update
through Yum
I've got an error said:
[sipxcha...@voip root]$ sipxconfig.sh
could not change directory to "/roo
To: chalek...@gmail.com; sipx-users@list.sipfoundry.org;
aidils...@prawedanet.co.id
Subject: RE: [sipx-users] Unable to get 4.0 started after update through Yum
Shutdown the rest of sipx
service sipxecs stop
then try to start sipxconfig only (no arguments)
sipxconfig.sh
If it comes up, are you ab
ackup before the upgrade?
-Original Message-
From: "Boy Aidil Sjam"
To: Charles
To:
Sent: 5/2/2009 4:04:45 AM
Subject: Re: [sipx-users] Unable to get 4.0 started after update through Yum
Just an update, I was able to manually update sipxsupervisor by running yum
install sip
: [ OK ]
Starting httpd:[ OK ]
Can someone help?
-Original Message-
From: Boy Aidil Sjam [mailto:aidils...@prawedanet.co.id]
Sent: Saturday, May 02, 2009 2:45 PM
To: 'Charles'; 'sipx-users@list.sipfoundry.o
I've got the same issue here after upgrade sipx 3.10.3 to sipxecs 4.0 using
yum
[r...@voip ~]# service sipxecs start
/usr/libexec/sipXecs/setup.d/011_sipx-config-httpd-access: line 16:
/etc/sipxpbx/httpd-sipxchange-common.conf: Permission denied
Checking bootstrap setup:
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