On 13 nov 2009, at 02.41, Matt White wrote:
> Your best bet for now is to simply disable gruu on the endpoints.
If I only could, that would've solved my problem a long time ago... Tandberg
won't let me do that though...
Best regards,
/Staffan
--
Staffan Kerker
mail/sip/xmpp: staf...@kerker.se
On Thu, 2009-11-12 at 13:10 -0800, IT Services wrote:
> One of our offices has only one network drop available for the 5 phones
> in that room. I set up an 8-port hub and started connecting phones to
> the hub. The first two phones were able to boot up successfully (getting
> the IP address via DHC
I think anyone responding would need to know who/what is providing
DHCP... SipX box or something else? I wouldn't think a hub vs a switch
would make in difference regarding what you're seeing. Any chance you
accidentally have two DHCP servers running?
-Original Message-
From: sipx-users
Your best bet for now is to simply disable gruu on the endpoints.
-m
>>> Staffan Kerker 11/12/09 9:49 AM >>>
Hi
I'm just wondering if this bug is being addressed in new releases?
Are there any workarounds, like disable the GRUU support in SipX, to make
endpoints that support GRUU work (by not
I didn't want to start another thread but have been asking if anyone using
these might
know if the 4.x firmware running on the used mediant 2k I bought would work
with sipx.
Mike
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On Wed, Nov 11, 2009 at 11:36 PM, li...@grounded.net wrote:
> > when you ran opensbc IT was the NAT traversal device instead of pfsense
> > is all.
>
> One very big difference is that now I'm using pfsense, which is a firewall.
> The only
> option I seem to have is pfsense in multi wan setup since
Hi all:
One of our offices has only one network drop available for the 5 phones
in that room. I set up an 8-port hub and started connecting phones to
the hub. The first two phones were able to boot up successfully (getting
the IP address via DHCP and registering with sipx). But it seems that
any p
Oops. Sorry. Duly noted.
So when can you fix t38modem so we can add faxing? Just kidding!
On Thu, Nov 12, 2009 at 1:49 PM, M. Ranganathan wrote:
> On Thu, Nov 12, 2009 at 1:46 PM, Tony Graziano
> wrote:
> >
> >
> > On Thu, Nov 12, 2009 at 1:36 PM, mkitchin.pub...@gmail.com
> > wrote:
> >>
> >
On Thu, Nov 12, 2009 at 1:46 PM, Tony Graziano
wrote:
>
>
> On Thu, Nov 12, 2009 at 1:36 PM, mkitchin.pub...@gmail.com
> wrote:
>>
>> I think I understand most of the concepts in play here, but I'm hoping
>> someone can recommend some specific devices, designs, ideas, best
>> practices etc.
>> We
On Thu, Nov 12, 2009 at 1:36 PM, mkitchin.pub...@gmail.com <
mkitchin.pub...@gmail.com> wrote:
> I think I understand most of the concepts in play here, but I'm hoping
> someone can recommend some specific devices, designs, ideas, best
> practices etc.
> We are planning on moving from a 2 PRI base
I think I understand most of the concepts in play here, but I'm hoping
someone can recommend some specific devices, designs, ideas, best
practices etc.
We are planning on moving from a 2 PRI based Nortel Norstar to Sipx
4.0.2. It is VOIP provided directly from Verizon over a Verizon DS3.
1) We
That sure sounds like my problem. I've installed the patch, restarted, and
had one successful call afterwards. I've asked the company operator to keep
me posted, but it looks like this fixed the issue. Thanks Tony!
Bruce
On 11/12/09 10:10 AM, "Tony Graziano" wrote:
> Please ensure you've patch
Unfortunately there is no fix for this other than submitting a bug to
Digium and having it ignored. The SIP stack in asterisk is pretty
shoddy, even in 1.6 as I currently see this problem as well. What you
can try to do is run it through the SBC (sipXbridge) as that will water
the SIP down enou
*IMPORTANT : If you are installing the pre-4.0.4 ISO image, please shut down
SipXbridge and then apply this patch:*
Please unzip in some empty directory in your file system and read/follow the
directions in README.
http://track.sipfoundry.org/secure/attachment/22556/patch20.zip
This patch fixes
Please ensure you've patched sipxbridge. The newest patch is on the wiki.
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: 434.984.842
On Thu, 2009-11-12 at 11:15 -0600, Gabe Casey wrote:
> I am having some issues using Asterisk as a PRI gateway with Sipxecs.
> For the most part it works for inbound and outbound calling however
> when a call is received on a PRI channel and then send to a SipXecs
> extension which has a forwarding
On Thu, 2009-11-12 at 10:40 -0500, Jhony Perez wrote:
> Hello everyone,
>
> I'm installing SipX 4.02 single CD install on a Dell PE R410 server
> with 4GB, 1066MHz, Dual Ranked UDIMMs memory, Intel Xeon E5502, 1.86 GHz
> processor and 2 SAS 146GB 15K RPM HHDs.
>
> When I boot from the disk, i
I am having some issues using Asterisk as a PRI gateway with Sipxecs. For the
most part it works for inbound and outbound calling however
when a call is received on a PRI channel and then send to a SipXecs extension
which has a forwarding rule to ring the extension and a mobile device at the
sa
On Thu, 2009-11-12 at 15:40 +, shaikh basha wrote:
> I tried a lot to know whether SIPXecs has built in VCC server (Voice
> Call Continuity)? I found on features list that VCC application. I
> wanted to clarify whether it supports to VCC enabled handsets such as
> nokia e52?
It should interop
On Thu, 2009-11-12 at 15:49 +0100, Staffan Kerker wrote:
> Hi
> I'm just wondering if this bug is being addressed in new releases?
http://sipx-wiki.calivia.com/index.php?title=When_will_release_X_be_available%3F&action=submit
Not on the road map at the moment...
> Are there any workarounds, lik
Dear All,
Good Morning,
I tried a lot to know whether SIPXecs has built in VCC server (Voice Call
Continuity)? I found on features list that VCC application. I wanted to clarify
whether it supports to VCC enabled handsets such as nokia e52?
Secondly, does SIPXecs provide free hosted sip servic
Hello everyone,
I'm installing SipX 4.02 single CD install on a Dell PE R410 server
with 4GB, 1066MHz, Dual Ranked UDIMMs memory, Intel Xeon E5502, 1.86 GHz
processor and 2 SAS 146GB 15K RPM HHDs.
When I boot from the disk, it goes to the SipX boot screen, I hit enter
and it start to run th
Ah, a license thing. But yes the debugging is powerful.
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 Contra
Hi
I'm just wondering if this bug is being addressed in new releases?
Are there any workarounds, like disable the GRUU support in SipX, to make
endpoints that support GRUU work (by not using GRUU in the first place)?
Best regards,
/Staffan
--
Staffan Kerker
mail/sip/xmpp: staf...@kerker.se
"
Probably should have something like that in the Wiki under the Polycom
page.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: Wednesday, November 11, 2009 12:46 PM
To: Sipx-users list
Subject: [sipx-users] sipx to harw
Thanks, Scott.
I filled a new issue for this problem.
On Thu, 2009-11-12 at 06:58 -0500, Scott Lawrence wrote:
> On Thu, 2009-11-12 at 11:36 +0200, an...@iguanait.com wrote:
> > Hi,
> >
> > we have installed Sipxecs 4.0.2-016420 on Centos 5.
> >
> > It is not behind NAT and it is configured pro
On Thu, 2009-11-12 at 11:36 +0200, an...@iguanait.com wrote:
> Hi,
>
> we have installed Sipxecs 4.0.2-016420 on Centos 5.
>
> It is not behind NAT and it is configured properly.
>
> We are facing a problem with one of the ITSPs that we have added as a
> Gateway. The ITSP is Telsome (telsome.com
Hi Tony,
As all the signalling is working I don't think there are any issues with
DNS/SipX.
Only the media part is broken, although RTP streams are flowing to and
from the Patton.
BTW: I don't use the WAN, it is not connected, this is just the default
config for the WAN port.
Thanks for the
Hi,
we have installed Sipxecs 4.0.2-016420 on Centos 5.
It is not behind NAT and it is configured properly.
We are facing a problem with one of the ITSPs that we have added as a
Gateway. The ITSP is Telsome (telsome.com).
Registration process is ok. We are able to register to the ITSP, but
whe
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