A wiki page would be great.
Tony Graziano wrote:
> How about a wiki page?
>
> >>> "Jermaine Pinder" <[EMAIL PROTECTED]> 10/27/2008 3:40:08 PM >>>
> I have lots of documentation on OpenSBC especially how to get it
> working with SIPX and Bandwidth.com.
>
> I even have firewall scripts, Traffic Sha
Anybody know a trick to get the voice mail to not record 5 min messages
all the time? I think there may be an issue with the audiocodes
recognizing the call has ended.
There sipx server is 192.168.10.6 and the audiocodes is 192.168.10.7
Here are the ini settings.
;**
;** Ini File *
I have about 30 of these phones with the Standard version of the firmware
and all register without issue and work fine until..
...you hit 20 minutes on a call and the call automatically drops. I have
been able to reproduce this with several phones and it is consistent.
I have many Polycom 430s an
On Mon, 2008-10-27 at 15:33 -0400, Jermaine Pinder wrote:
> We are experiencing low MOS score, Jitter, and packet loss readings
> consistently, especially when they go to retrieve their voice
> messages. All other calls outside are working fine, MOS scores are
> around 4.4 with no packet loss and l
How about a wiki page?
>>> "Jermaine Pinder" <[EMAIL PROTECTED]> 10/27/2008 3:40:08 PM >>>
I have lots of documentation on OpenSBC especially how to get it
working with SIPX and Bandwidth.com.
I even have firewall scripts, Traffic Shaping and Upper Registration
documentation that would be useful
We are experiencing low MOS score, Jitter, and packet loss readings
consistently, especially when they go to retrieve their voice messages.
All other calls outside are working fine, MOS scores are around 4.4 with
no packet loss and low Jitter (under 20ms).
Im using VQmanger as a test tool.
Any i
I have lots of documentation on OpenSBC especially how to get it working with
SIPX and Bandwidth.com.
I even have firewall scripts, Traffic Shaping and Upper Registration
documentation that would be useful to users who want to OpenSBC.
Creating a new mailing list would be great.
__
Hi Tony
In my inGate setup
eth0 private ipaddr of lan INSIDE 192.168.10.0/24
eth1 is Internet ITSP
eth2 private ipaddr is where SIPX_LAN lives 192.168.20.2/24
Internal phones are on eth2 dhcp/ddns and can place and receive calls presently.
Both STUN and Remote NAT Traversal in "Remote SIP Conne
No tricks.
This assumes ETH0 is on the network where sipx is installed.
ETH0 has to have two IP's bound to it. The ALIAS (second IP) needs to
be the one used by sipx for siptrunk/gateway addressing.
No change is necessary on ETH1.
This way registration occur on the main private IP, but trun
Hi
Has anyone gotten x-lite 4.0 to register from remote location through inGate =>
sipX? I have gotten it to ring on remote laptop from an AT&T phone but will
not register from remote laptop. What's the trick?
r___
sipx-users mailing list
sipx-users
Hi,
I run an istance of sipxecs in HA configuration (sipx version 3.10.1),
when a phone is registered in slave server and makes a call to another
phone, I can't see the "in progress calls" on "Call Detail Records" page
and the call details aren't recorded in "historic" on "calls" page.
Someone
On Mon, 2008-10-27 at 12:44 +0900, Joegen E. Baclor wrote:
> Hi,
>
> I am sure the bridge is equally as good if not a better choice than
> OpenSBC. Having direct configuration link to sipxConfig, I wouldn't
> doubt the bridges edge over OpenSBC in that regard. However, there
> might be indee
Hi,
first of all, i would like to thanks to you for this discussion. I'm
sure that it will be useful for many people.
I want to ask why do i need to set 2 gateways for the same provider with
different dialing plans? I know that i can create different dial plans
(for local and for long distance
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