I am using snom 730 phone which is registered with sipx server. It
appears that sip trunc registration between sipx and ITSP was
successfull. My sipx server is behind nat firewall. When I place a
phone call from outside (pstn), the snom 370 is ringing and displays
caller id. When I pick up the
> And at this point I think only DID faxing via t.38 is being addressed. This
> means if your provider doesn't support t.38 you will want to get trunks from
> one that does support it (or go through a PRI gateway).
We use the mediant 2K with PRI's and route faxes to a fax server. I would love
to
And at this point I think only DID faxing via t.38 is being addressed. This
means if your provider doesn't support t.38 you will want to get trunks from
one that does support it (or go through a PRI gateway).
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984
Michael B. has t38 this working on his systems, I'm in the process of
pulling in his changes into SF upstream as we speak. You can add
yourself to watch list for this issue to make sure you do not miss
anything
http://track.sipfoundry.org/browse/XX-8645
There were few release engineering changes
I have seen a lot of mention of faxing on sipx so I decided to take a look to
see what's up.
I am unable to confirm if this is something coming in a new version or just the
usual ongoing chatter about it. Searching sipfoundry doesn't reveal much. Can
someone confirm if this is something new com
New features worth noting:
5: Added the ability to trigger a reboot (or configuration update) from the
microbrowser. E.g.
62259: Phones now display the Call Forward destination on Idle Display.
23335: Configuration parameter values can now be updated at run-time.
48138: SoundPoint IP320,
5080 for trunking. 5060 tcp/udp for remote users and 3-31000 udp for
media.
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.
On 10/7/10 3:12 PM, Roman Gelfand wrote:
If I am not mistaken, I have to publish ports 5080/udp and
3-31000/tcp on the wan.
5060, udp/tcp if you support inbound sip: url calls and/or remote users.
5080/udp/tcp for ITSP's
3-31000 UDP.
note: turn off all sip helpers, alg, etc.
you MUST
http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip550.html looks
like the cat's out of the bag. Time to get started on 3.3.1 interop :)
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If I am not mistaken, I have to publish ports 5080/udp and
3-31000/tcp on the wan.
Is that correct?
Thanks in advance
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Can anyone send me some pointers for getting a Polycom Spectralink
8020 with 4.2.1? There doesn't seem to be a lot out there on this.
For handsets so far, I have only done Polycom 450 and 550s. The fact
that I know so little about how to setup a handset is a testament to how
well Sipx manages
Welcome to the wild side George!
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Martin
Steinmann
Sent: Thursday, October 07, 2010 6:58 AM
To: 'Discussion list for users of sipXecs software'
Subject: [sipx-users] Welcome to George Nicula
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rene Pankratz
[rene.pankratz.l...@iant.de]
I have a dialplan rule with several gateways associated (a VoIP provider and a
patton ISDN gateway).
Now, if I
(in my instance we were able to configure the ACL/Dialplan in the AC gateway
to accept the calls without SAS altogether, though we were using an FXO
gateway)
On Thu, Oct 7, 2010 at 11:22 AM, Tony Graziano wrote:
> The polycom has an option in it to allow already. Phone>Dial Plan
> (advanced)> Em
The polycom has an option in it to allow already. Phone>Dial Plan
(advanced)> Emergency Routing.
That part explains itself. Just change the # from 911 to whatever you want
(cell phone to test). Then make sure the gateway (AudioCodes) is setup
properly. There's a wiki page for that;
http://wiki.si
That sounds like a great idea Tony. Can you expand a little on how this is
done?
In case it's relevant we use Polycom Soundpoint 330/650's and Audiocodes
Mediant 1000 gateways.
Regards
Huw
--
Huw Wyn Jones
Systems Administrator
Coleg Meirion Dwyfor
huw.jo...@meirion-dwyfor.ac.uk
-
And you can put any number in you want to test with... If the phone/gateway
has that option
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.842
Depending on the phone, you can use the option to have the phone call the
gateway directly and bypass sipx altogether. That is the desired method we
use here.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Tele
On Thu, Oct 7, 2010 at 9:58 AM, Martin Steinmann wrote:
> Would like to welcome George to the sipXecs developer team. George has been
> active with sipXecs for quite some time and we are very fortunate to have
> him on board. Looking forward to a lot of great patches and a lively
> discussion on
In the US (911) I don't know of a way to thoroughly test it with out actually
doing it. I call the police department on their non emergency line first, and
schedule the test during a set time.
-Original Message-
From: Huw Wyn Jones
Sender: sipx-users-boun...@list.sipfoundry.org
Date: Th
Hi folks.
We had a problem today when a member of staff needed to call the emergency
services. The call failed on one phone, but succeeded (thank goodness) on
another. As a result I'm now rechecking our dial plans to see why the first
call might have failed. My question is how do you check if t
Done.
http://track.sipfoundry.org/browse/XX-9064
On 10/7/2010 1:11 AM, Michael Picher wrote:
I can confirm this Matthew. Please open a ticket on it.
Thanks,
Mike
On Wed, Oct 6, 2010 at 1:03 PM, Matthew Kitchin (public/usenet)
mailto:mkitchin.pub...@gmail.com>> wrote:
I upgraded from
Thanks. I definitely ran into some of the issues you did. I think I
tried so many things the first time around, I wasn't sure what all
worked and what didn't. I cleared everything out and started from
scratch a few days ago, and got it to work.
On 10/7/2010 8:34 AM, Geoff Van Brunt wrote:
> I
Would like to welcome George to the sipXecs developer team. George has been
active with sipXecs for quite some time and we are very fortunate to have
him on board. Looking forward to a lot of great patches and a lively
discussion on the lists.
Welcome George!
--martin
___
I just did this yesterday in fact. I never could get the web gui
working, except for the CA certs. That required exporting in base-64
format and then changing the file extension to crt from cer, otherwise
they won't upload. I haven't gotten around to checking the tracker if an
issue has been create
On Thu, Oct 7, 2010 at 2:39 AM, Rene Pankratz
wrote:
> I also thought about creating a feature request in the tracker.
> It would be really helpful if this option would be set within the
> preconfigured dhcp server and I cannot see a problem with other devices in
> the network.
http://track.sipfo
Hi,
Regarding http://track.sipfoundry.org/browse/XX-8673, anybody seen this
lately?
I can't reproduce this error on my build machine. I'm running a
sipXconfig 4.3.0-019022
Regards
Cristi
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List
Hello,
I have a dialplan rule with several gateways associated (a VoIP provider and
a patton ISDN gateway).
Now, if I call someone who desn't answer or is busy I see that sipx calls
the callee twice.
Even if the VoIP provider sends a 486 to sipx, sipx switches to the patton
gateway for a second t
On 6 okt 2010, at 23.45, Worley, Dale R (Dale) wrote:
> I now have time to look into this.
Great!
> It looks like this is a problem with the remote NAT traversal feature. It's
> possible that it isn't specific to GRUU processing. I will talk to our
> expert on NAT Traversal.
OK, just let m
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