On Fri, 2008-08-08 at 09:15 +0800, Dean Hiller wrote:
> What is suggested then on routing calls through my wan port? Is there
> any cheap kind of equipment I can buy. I am trying to setup a network
> with phones -> sipx -> router to wan -> sip trunking provider service
> as I have software that
What is suggested then on routing calls through my wan port? Is there any
cheap kind of equipment I can buy. I am trying to setup a network with
phones -> sipx -> router to wan -> sip trunking provider service as I have
software that I want to launch a product with.
I have been also trying to te
SIP is not exactly simple. You need to understand why it uses DNS, the
options it uses for DHCP and so forth.
It is "rolled" into a nice package to deliver all the basic needs into a
single CD because a lot of admins will run it on a separate IP
network/vlan or in a lab environment to learn the te
I am not sure I know how to do all that with dns, dhcp as that seemed like
the harder route. isn't that bad to have two dhcp machines on your network
as my linksys is doing dhcp? Iam pretty sure I can turn dhcp off on the
linksys but if I do that, and sipx is assigning ip addresses, will my
netw
On Thu, 2008-08-07 at 18:51 -0300, Melcon Moraes wrote:
> :(
>
> No can do. I can't see a link to open a new issue.
>
> I'll open it as soon as I have access to do so.
Done.
--
Scott Lawrence tel:+1.781.229.0533;ext=162 or sip:[EMAIL PROTECTED]
sipXecs project coordinator - SIPfoundry http
Hello,
SipX 3.10.2 works very well for me (thanks again for the help), so I
decided to test out the Aug 1 build 3.11.4. Everything looks OK except a
freeswitch user not being found on startup. Right out of the gate there
is a nat traversal rules job failure upon initial web login. I realize
th
:(
No can do. I can't see a link to open a new issue.
I'll open it as soon as I have access to do so.
[]'s
MM
On Thu, Aug 7, 2008 at 6:24 PM, Picher, Michael
<[EMAIL PROTECTED]> wrote:
> I would...
>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:sipx-users-
>> [EMAIL PROTECTE
Back up the train...
Why don't you lab this thing on some private addressing on a regular
network... you have my head spinning here with yours, and your friends
and you both have things in dmz's.
Walk before you run. Stay away from nats, dns mess and firewalls until
you know you have a
I would...
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:sipx-users-
> [EMAIL PROTECTED] On Behalf Of Melcon Moraes
> Sent: Wednesday, August 06, 2008 8:53 PM
> To: sipX users
> Subject: Re: [sipx-users] ACD Reports: inconsistency on Agent
> ActivitySummary numbers
>
> Hi Damian,
On Wed, 2008-08-06 at 07:49 +, Panagiotis Bekiaris wrote:
> > I have installed version SIPx 3.10.2.
> > I have tried to keep the configuration pretty simple. 1 polycom sipphone
> > and
> 1
> > GW. The GW is propertly configured and works 100%.
> > I can call the voicemail from the phome but
On Thu, 2008-08-07 at 15:18 +0100, Gabor Paller wrote:
> Indeed, the cause was the lack of TCP listening point on my side.
> Creating a TCP ListeningPoint and adding it to the SipProvider instance
> solved the problem.
The alternative solution would be to add ";transport=udp" to the contact
URI th
On Aug 6, 2008, at 10:28 PM, Charles P McParland wrote:
> I have dug into my ISO 3,10,1 and 3.10.2 problem and
> found the following:
>
> -Burned CD's that do not boot on the target PC (a VIA
> EPIA 8000 mini ITX) DO boot on several other PC's.
>
> -Centos OS ISO images boot on both the above tar
Indeed, the cause was the lack of TCP listening point on my side.
Creating a TCP ListeningPoint and adding it to the SipProvider instance
solved the problem.
Thanks for the quick and useful response.
Regards,
Gabor
PS: What is the Java version SipX depends on? :-)
-Original Message-
From
On Thu, 2008-08-07 at 13:01 +0100, Gabor Paller wrote:
> Dear list,
>
> I have just upgraded to 3.10.2 from 3.10.1 and I ran into an issue with
> the new SIP proxy. I have a SIP application based on JAIN-SIP (currently
> using version 1.2.76). The application registers on behalf of a SIP
> user,
Dear list,
I have just upgraded to 3.10.2 from 3.10.1 and I ran into an issue with
the new SIP proxy. I have a SIP application based on JAIN-SIP (currently
using version 1.2.76). The application registers on behalf of a SIP
user, accepts INVITEs and initiates calls when incoming INVITEs arrive.
Wi
On Thu, 2008-08-07 at 12:25 +0300, otto wrote:
> I'm not understood , where should I look in sipx or asterisk configuration?
> If this is not an error how we should configure asterisk or sipx for
> compatibility?
If you can't find a version of Asterisk that does not have that bug,
then it just wo
On Thu, 2008-08-07 at 11:26 +0800, Dean Hiller wrote:
> My friend is having a different issue. His sipx is behind a
> router/NAT as well, but he actually has DNS sipx.x.com pointing to
> that router and then has sipx in the DMZ(I am doing something
> similar).
>
> For some reason, when sipx
I'm not understood , where should I look in sipx or asterisk configuration?
If this is not an error how we should configure asterisk or sipx for
compatibility?
Otto
-Original Message-
From: Paweł Pierścionek [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 05, 2008 11:08 AM
To: sipx-use
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