VPN?
On Tue, Dec 13, 2011 at 1:23 AM, Spiny Tel spiny...@gmail.com wrote:
Dear,
I need a solution for VOIP blocked country, Is there any B2BUA or proxy
for that purpose.
SIPx supports like that?
Thanks
___
sipx-users mailing list
Created http://track.sipfoundry.org/browse/XX-9995 to get this improved.
Paul
Worley, Dale R (Dale) dwor...@avaya.com wrote on 12-12-2011 18:29:02:
From: Michael Picher [mpic...@ezuce.com]
I think the default install has always installed with weight 0 SRV
records.
Well, it
Yeah. It can be called as VPN solution to. Our SIP client is based on open
source PJSIP (www.pjsip.org). sipXecs software supports that?
On Tue, Dec 13, 2011 at 3:47 PM, Michael Picher mpic...@ezuce.com wrote:
VPN?
On Tue, Dec 13, 2011 at 1:23 AM, Spiny Tel spiny...@gmail.com wrote:
Dear,
As far as I know, yes.
On Tue, Dec 13, 2011 at 6:15 AM, Spiny Tel spiny...@gmail.com wrote:
Yeah. It can be called as VPN solution to. Our SIP client is based on open
source PJSIP (www.pjsip.org). sipXecs software supports that?
On Tue, Dec 13, 2011 at 3:47 PM, Michael Picher
Can you please provide me some demo's or quick links to do some testings?
Below is my targeted scenario
SIP Client - (encrypted SIP + RTP ) Encryption Server --
SIP Registrar PSTN
On Tue, Dec 13, 2011 at 5:30 PM, Michael Picher mpic...@ezuce.com wrote:
As far as I know,
what are you talking about?
get your sip client from wherever... download and install sipXecs from
http://download.sipfoundry.org and install it on your own machine...
On Tue, Dec 13, 2011 at 6:53 AM, Spiny Tel spiny...@gmail.com wrote:
Can you please provide me some demo's or quick links to
I was wondering if anyone had experience with SIPX installed alongside
the LSI MegaRaid MSM utilities. The configuration under consideration
is a re-purposed ASUS RS500 server with a ASUS Pike 1064E raid
controller based on LSI MegaRaid.
What we are attempting to achieve is the installation
I've not seen any inquiries to the list about this that I can recall.
I'd do a base CentOS install, get your Raid controller running and then do
a yum based install of openUC.
Mike
On Tue, Dec 13, 2011 at 8:43 AM, Black, Dave (CallPoint Canada)
dave.bl...@callpointcanada.ca wrote:
I was
I think what he is trying to do is use SIP to/from a country which has
blocked SIP in general. You can do this with most any platform as long as
you can establish an encryted tunnel or use a proprietary encryption method
(like CBCOM) which is vendor specific. You will be best suited to
It's really a matter of whether or not the utilities are supported in
Linux. I know there are ways to install the Dell management utilities in
Centos, we've done that. The question you are asking is plainly a Linux
question. I am pretty sure they are not supported on certain customized
versions of
On 12/13/2011 8:43 AM, Black, Dave (CallPoint Canada) wrote:
I was wondering if anyone had experience with SIPX installed alongside
the LSI MegaRaid MSM utilities. The configuration under consideration
is a re-purposed ASUS RS500 server with a ASUS Pike 1064E raid
controller based on LSI
We have an unusual problem with one of our DIDs not working correctly and
cannot seem to figure out. If someone have experienced this before or can
point me in the right direction, it would be appreciated. We have the most
current sipx version loaded. PBX has been restarted. All other DIDs
Have you tried your PSTN provider at the CO level? To me it seems to be a
DID issue.
Solomon
From: Ly Tran ly.t...@synaptyk.com
To: sipx-users@list.sipfoundry.org sipx-users@list.sipfoundry.org
Date: 12/13/2011 01:14 PM
Subject:[sipx-users] Incoming DID Not Working
Sent by:
You would need a sip trace to make sure it sends the invite on port 5080
and from the expected ip address or gateway with your provider.
Then we would want to make sure it does a proper reinvite for t.38.
I suspect it does not come from the same provider gateway but the best
thing to do is to
Let me elaborate...
I see the same problem with providers who do not send rtp on from the same
address the invite comes from. This usually becomes a problem especially
when you use more than one wan connection.
On Dec 13, 2011 1:43 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
You would
Thanks Tony. I did a siptrace and could not see any traffic coming in when
initiating an external call to that DID.. not even an Invite. Called ITSP and
told them this and to check configuration again for this number. Problem now
fixed. T.38 was enabled, but the DID was not assigned to our
No Invite no ring. Glad it was that easy to solve.
On Dec 13, 2011 4:09 PM, Ly Tran ly.t...@synaptyk.com wrote:
Thanks Tony. I did a siptrace and could not see any traffic coming in
when initiating an external call to that DID.. not even an Invite. Called
ITSP and told them this and to
I have a Sipx system set up with a sip trunk using the internal sipxbridge
I have Gateway: Broadvox /SIP trunk /Default Caller ID set to 5551231234
When a user maks an external call the CLID Name is sent out as Sipxbridge with
the number 5551231234
The User has no CLID settings configured
When
CNAM (Caller ID name) is controlled by the ITSp or carrier. The only time I
see CNAM not being controlled by the carrier is when the calls are between
two accounts on the same carrier.
CLID and CNAM are two different pieces of the caller id display.
I would ask the carrier to set the CNAM to the
I just got off the phone with ITSp
They can populate the CNAM, but, that will only populate to the PSTN (land
lines)
Any SIP calls, and calls to Comcast users will receive the CNAM the Sipx sends
out
If I want it to work for all carriers, I need to send it
I'm back to my original question, can
Wow. Who is the itsp? I've never seen one that reads the sip header wrong
like that.
On Dec 13, 2011 8:13 PM, Ken Ridley k...@federico.net wrote:
I just got off the phone with ITSp
They can populate the CNAM, but, that will only populate to the PSTN (land
lines)
Any SIP calls, and
And what I mean by that is that the carrier is interpreting the sip user
agent name as the caller id name.
I don't think there's anything wrong with the way that sipx is sending the
caller id name. I think the issue is with how the switch at the other end
is parsing the hatter improperly.
On Dec
Try mucking with the CallerID settings on the gateway (left side menu).
Click on Advanced, you'll see more options.
Mike
On Tue, Dec 13, 2011 at 8:28 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
And what I mean by that is that the carrier is interpreting the sip user
agent name as
I'd like to hear who the ITSP is. I'd also like to see if somehow the
header is parsed differently using the advanced setting for the caller id
name. The sip header sends the username (not sipxbridge) now. So it would
be interesting to see if the results vary in how the ITSP parses this.
On Tue,
Well, I do know with voip.ms if I call another voip.ms customer my username
gets passed to the other voip.ms customer.
Not the case however if I dial over to another provider.
Mike
On Tue, Dec 13, 2011 at 8:56 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
I'd like to hear who the
Calling Name for traditional land line and wireless carriers is determined
by a database lookup by the carrier terminating the call. It is based on
the Caller ID. It is a fee based service in which both the originating and
terminating carriers must agree to participate. Almost all land line
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