Indeed you are right it was an IM id for another user.
I appreciate the help
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Center Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Cell: 615-852-5015
Fax: 615-628-5698
Email:gca...@franklinamer
I've been told that Broadvox will be rolling out their new user portal in
early 2011, which will have the features you are looking for.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Gary Luca
Sent: Monday, November 01, 2010 12:01 PM
To
look at the aliases file directly
cat /var/sipxdata/sipdb/aliases.xml
im betting its assigned somewhere
On Mon, Nov 1, 2010 at 5:23 PM, Gabe Casey wrote:
> I am getting the error The user ID or alias "1000" duplicates an existing
> alias for a user or service
> I have re indexed checked both th
I am getting the error The user ID or alias "1000" duplicates an existing alias
for a user or service
I have re indexed checked both the users , IM and conference and dialplan i can
not find any instance of this
I have even run queries on the database to find it. Nothing.
How can i move past
you need a siptrace in this instance to see whats up.
On Mon, Nov 1, 2010 at 5:20 PM, Sven Evensen wrote:
> Since calls to VM are working, it seems the license is installed
> correctly. Unless your internal
>
> calls are not using G729.
>
>
>
> A look at wireshark will quickly show what is wrong
Since calls to VM are working, it seems the license is installed correctly.
Unless your internal
calls are not using G729.
A look at wireshark will quickly show what is wrong here. If it is a codec
problem, you will
normally see "488 Not acceptable"
Sven
its been discussed, but like anything in the roadmap, its not committed to
at this time and the discussion has not taken place yet as it is not
scheduled to until after 4.4 is release and checked out.
On Mon, Nov 1, 2010 at 4:55 PM, Roman Gelfand wrote:
> Judging by your response, it is in the p
did you verify the codec is running and license is available using the
fs_cli?
On Mon, Nov 1, 2010 at 4:47 PM, James Black wrote:
> Hi Sven,
>
>
>
> Thanks, have tried that too. No luck. It’s almost like the call is not
> transferred/answered by voicemail ITSP, or its answered and then it hangs
Judging by your response, it is in the plans. Any idea when?
On Mon, Nov 1, 2010 at 4:28 PM, Tony Graziano
wrote:
> No, just the rtp range at this time.
>
> Tony Graziano, Manager
> Telephone: 434.984.8430
> Fax: 434.984.8431
>
> Email: tgrazi...@myitdepartment.net
>
Hi Sven,
Thanks, have tried that too. No luck. It's almost like the call is not
transferred/answered by voicemail ITSP, or its answered and then it hangs up.
The reason that I have to use G.729AB is that it is the only Codec the ITSP
uses.
Kind regards
James Black
From: Sven Eve
yes sorry i verified that we are using apache as a proxy to the portal.
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Center Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Cell: 615-852-5015
Fax: 615-628-5698
Email:gca...@franklinamerican.co
No, just the rtp range at this time.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http:
I was refering to the NAT configuration where public port is 5060.
On Mon, Nov 1, 2010 at 4:25 PM, Roman Gelfand wrote:
> If I am not mistaken, I think it was mentioned before that remote
> registrations can only take place on port 5060. However, it appears
> that I could change that, including
If I am not mistaken, I think it was mentioned before that remote
registrations can only take place on port 5060. However, it appears
that I could change that, including rtp port range. Is that right?
Thanks in advance
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sipx-users mailing list
sipx-
i'm not an expert, but i think you have to setup a proxy and not a rewrite.
On Mon, Nov 1, 2010 at 3:27 PM, Gabe Casey wrote:
> is there a setting or ssl crt that needs to be adjusted when doing an
> external rewrite of the url to the end user portal from outside the lan.
> My goal is to allow ac
In preparation for wiki clean-up day on friday, I'm combining all the
pages into a simple space. The way it was organized into multiple
spaces was not conducive to maintenance. Unfortunately this changes
the URL for everything. There be a lot more changes, i just wanted to
give everyone a heads up
I've never had them go down. I say that while my bandwidth.com trunks have
gone down now again (2nd time in 6 months) using ip based.
Nice that you can have automated failover with voip.ms.
I'd pursue to use a t.38 provider as backup for primary voice and tro carry
fax calls.
On 11/1/10 3:42 PM, Todd Hodgen wrote:
I am a reseller of Broadvox, and a user of Voip.ms. I don't
recommend Voip.ms to customers except for labs, etc. as I find that
they are not totally reliable. They are very cost effective, if you
have a customer that does not sit at their phone all d
I am a reseller of Broadvox, and a user of Voip.ms. I don't recommend
Voip.ms to customers except for labs, etc. as I find that they are not
totally reliable. They are very cost effective, if you have a customer
that does not sit at their phone all day long, it works great, but if they
are on
is there a setting or ssl crt that needs to be adjusted when doing an external
rewrite of the url to the end user portal from outside the lan.
My goal is to allow access of the user portal outside the network, however the
browser is throwing errors and forcing a logout.
voicemail.example.com -
It's the only ITSP I've been happy with to date (I've tried 3 or 4). We are
running our long distance calls to them and have been very pleased with the
quality. We are averaging between 75-125 ld calls per day (approx. 4 hours).
I have only used them in a very limited capacity for inbound cal
Good for me so far. They use http://www.theplanet.com/ to host their servers
(at least in Texas where I am). The only thing you need to do is ask them if
the DID's you're looking to buy/port have the capability of outbound caller ID
name. Mine don't so they show up as "Bryan, TX" when I call som
Hello all,
I am planning to implement a new sipX solution for an organization. I set
one up for another organization about five months ago and used Broadvox as
the SIP trunk provider. I haven't really had any problems with them and the
cost is low. But one thing I don't like about them is that
This also assumes the ITSP allows or is configured to allow G729. I suspect
the ITSP is allowing it, otherwise the call should fail, but not all ITSP's
give appropriate error codes (not allowed, etc.).
On Mon, Nov 1, 2010 at 8:06 AM, Sven Evensen wrote:
> Hi James,
>
>
>
> Check the SIP trunking
Hi James,
Check the SIP trunking config, advanced settings.
The Permitted Codecs does not include G729 by default
Either leave the field blank, or add G729.
Sven
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.
As Tony mentioned, from a Polycom perspective you do need the Productivity
Pack (licensed per station), and you really want to have Soundpoint 450's or
better for the larger screens.
With the productivity pack and the LDAP look up the contacts are not stored
in the phone, they get resolved on the
Hi Arman,
I have subsequently installed with the
http://files.freeswitch.org/g729/fsg729-153-installer
The AA/Voicemail is now working for internal G729 Calls (over our vpn), however
when I get calls coming in from our ITSP, as soon as the call is supposed to
get answered, it is just cut
Hi,
Together with a colleague of mine, we reported a bug namely:
http://track.sipfoundry.org/browse/XX-8309
Since we really needed a fix, we started to fix it ourselves.
Could someone confirm that this fixes the problems and add this fix to a next
stable release?
The patch works for
Are you able to build a system from the latest snapshot rpm's to determine
if the issue is resolved up to the point when the gruu is not behind nat?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Secu
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