e
>
> On Fri, Sep 9, 2011 at 1:12 PM, Gary Luca wrote:
>>
>> And I have a new problem.
>>
>> Shared Line Appearance = fail
>>
>> The line registers no problem but all I get is a flashing red light on
>> the line key and no ability to make or receive
11 at 1:10 PM, Gary Luca wrote:
> I've BEEN configuring them manually. There is no support in sipx for
> SPA502G or SPA504G.
>
> Configured to use UDP = no incoming calls
>
> And I haven't a clue how I would go about dumping the configuration
> off the phone.
&
, Sep 9, 2011 at 12:25 PM, Gary Luca wrote:
>> Just for the hell of it I tried using TCP instead of UDP for SIP
>> signalling. And it works no problem.
>
> Can you configure the phone manually to do UDP and get inbound calls
> to work? If so, can you dump the configuration som
UPDATE:
Just for the hell of it I tried using TCP instead of UDP for SIP
signalling. And it works no problem.
What the crap?!
On Thu, Sep 8, 2011 at 9:07 PM, Gary Luca wrote:
> You live...you learn.
>
>
> On Thu, Sep 8, 2011 at 8:58 PM, Michael Picher wrote:
>> I gues
You live...you learn.
On Thu, Sep 8, 2011 at 8:58 PM, Michael Picher wrote:
> I guess next time you'll have your ammunition and just say 'no, there isn't
> anything better that works as well...'
>
> On Thu, Sep 8, 2011 at 8:51 PM, Gary Luca wrote:
>>
>&
s.
> Of course that doesn't help you at the moment. Nor can I other than to
> suggest you should perform a trace and post that in case some other person
> who likes pull their hair out has figured it out.
> Mike
> On Thu, Sep 8, 2011 at 4:50 PM, Gary Luca wrote:
>>
>&g
Couple of days ago I had someone trying to make international calls to
Israel. I tracked the source IP address. Registered to an Internet
provider in Palestine.
After that, I closed up port 5060 on the firewall. I don't need it right
now since I don't have remote workers and all ITSP calls goes
Hello all,
Don't know if anyone has encountered this issue yet. I've been
searching high and low all day and driving myself crazy trying to
figure out why my brand new Cisco SPA504G and SPA502G phones are able
to make outbound calls with no problem but absolutely will not receive
inbound calls.
On Behalf Of Tony Graziano
> Sent: Wednesday, June 15, 2011 2:05 PM
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] Server stopped responding
>
> the registrar log would yield easy to spot information if it were
> sipvicious.
>
> On Wed, Jun 15
vicious attacks.
>
> Additionally, if CPU had for some reason gone through the roof (top),
> of course, if that was unavailable, your vmware server stats would
> indicate as much.
>
> Finally, there are some more recent patches you might consider picking
> up via the repo (stab
Hello all,
My sipx server stopped responding for some reason. All users were
disconnected and unable to register. I rebooted the whole server and it
came back up.
I've taken a snapshot. What files can I send along that will help to
diagnose the issue? Give me specific locations and filenames
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
Please forgive me. I'm not much of a Linux person at all. I've done
everything up to "merge-logs". What exactly do I do now?
Thanks
On Wed, Oct 27, 2010 at 9:17 PM, Tony Graziano wrote:
> Yes. Set the proxy to debug log level.
>
> If you don't care about your logs do this:
>
> cd /var/log/si
Hello all,
I'm having an issue with getting DID numbers to route properly. I am using
Broadvox for SIP trunking. I have two phone numbers set up. One is the
primary number (the BTN or Billing Telephone Number). We'll call that
781-555-5000. The second number is a VDID (Virtual Direct Inward D
url" to have version 4.2.1 instead of 4.2.0 and then to
> "CTRL+X" and save, overwriting the same filename.
>
>
> Now do:
>
>
> yum clean all
>
> yum update
>
>
> ALWAYS a good idea to do a sipxbackup first!
>
>
>
>
>
> On Mon, O
o
> http://download.sipfoundry.org and get the new .repo file. The server
> name has changes from sipxecs to download. Replace the repo file in
> /etc/yum.repos.d with the one you just downloaded, and things should
> be back to working.
>
>
> Matt P.
>
> On Sat, Oct 2,
Hello all,
Please find the two attached documents. One is a copy/paste of my "updates
are ready to be installed" page and the other is the installation log. It
keeps failing for some reason and i'm not totally sure why. It seems like
sipX is trying to download the repositories but can't find th
Tony,
I'm using a SIP trunk through Broadvox with the default settings as shown
below in your email and have had no reported issues with dropped audio after
30 minutes. However they DID do some sort of tweak on their end back when
we set the trunk up because incoming calls would get killed if you
aziano <
> tgrazi...@myitdepartment.net
> > wrote:
>
> > If static nat isn't supported at the other end it will systematically
> > break...
> >
> >
> > Tony Graziano, Manager
> > Telephone: 434.984.8430
> > Fax: 434.
: 434.984.8426
> Fax: 434.984.8427
>
> Helpdesk Contract Customers:
> http://www.myitdepartment.net/gethelp/
>
> - Original Message -
> From: sipx-users-boun...@list.sipfoundry.org
>
> To: Gary Luca ; sipx-users@list.sipfoundry.org
>
> Sent: Thu Jul 15 19:05:52
Yes, indeed, Nate. That baby took a HUGE leap. haha
On Thu, Jul 15, 2010 at 7:05 PM, Nathaniel Watkins <
nwatk...@garrettcounty.org> wrote:
> Must be a big baby…
>
>
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org]
> cases. Registering is only a babystep.
>
>
> On 7/15/10, Gary Luca wrote:
> > I DID IT
> >
> > It is the STUPIDEST little thing.
> >
> > So in the phone group setup, I was specifying the outgoing proxy in the
> SIP
> > Servers screen. Just for
t; totally different one, but you must be able to both control it and update
> it).
>
> You are spinning your wheels, I'm afraid it's somewhat futile, you simply
> don't have a checklist that is properly prepared.
>
> On Thu, Jul 15, 2010 at 5:51 PM, Gary Luca wrote:
&g
cessary records. Though I
still don't see why this would be necessary if I'm specifying an Outgoing
Proxy. x-lite works without the external DNS entries, so why wouldn't the
Polycom 430?
Am I missing something?
On Thu, Jul 15, 2010 at 5:42 PM, Gary Luca wrote:
> There are no managed
; Can you put these phones on a vlan that relays dhcp requests to your sipx
> server? Not exactly what you are looking for, but that is how I handle it.
> -Original Message-
> From: Gary Luca
> Sender: sipx-users-boun...@list.sipfoundry.org
> Date: Thu, 15 Jul 2010 17:11:49
#x27;t even
> need the phone setup in sipx or need to have a user assigned to it.
> -Original Message-
> From: Gary Luca
> Sender: sipx-users-boun...@list.sipfoundry.org
> Date: Thu, 15 Jul 2010 16:59:44
> To:
> Subject: Re: [sipx-users] Polycom IP 430 won't regist
t; NAT (which I doubt).
> > >
> > > So once you get DNS setup where the phones will
> > > register you should try making a call and if you get no (or
> > > one-way audio), you should stop and address the firewall piece.
> > >
> > >
ece.
> >
> > If it were me (and it is not), I would see if I
> > could set the verizon modem to "bridged" mode and put a
> > compatible firewall in. I say this because in looking through
> > the fios devices of that model, I see nothing which
&g
bridged" mode and put a compatible firewall in. I say this because
>> in looking through the fios devices of that model, I see nothing which
>> encourages me that this is something that has the guts to work in the way
>> that is needed, but is fine for a remote site.
>>
&g
fora4senate.com
Password: *
Display name: Gary Luca
Authorization name: [blank]
Register with domain and receive calls: [checked]
Send outbound via: Proxy - Address: *myhostname*.no-ip.org
Dial plan: #1\a\a.T;match=1;prestrip=2;
Topology
IP Address: Use l
, 2010, at 8:13 PM, Scott Lawrence wrote:
> On 2010-07-14 17:16, Gary Luca wrote:
>> I already posted this on the forum, but haven't gotten any replies
>> yet. I know i'm not supposed to send to the list as well, but i'm not
>> sure if it got sent out. Now th
ty and Control Systems Helpdesk:
> Telephone: 434.984.8426
> Fax: 434.984.8427
>
> Helpdesk Contract Customers:
> http://www.myitdepartment.net/gethelp/
>
> - Original Message -
> From: Gary Luca Jr
> To: Tony Graziano
> Cc: sipx-users@list.sipfoundry.org
>
Tony. Thank you. I'm going to try re-provisioning tomorrow and downgrading back
to 3.1.3. I was going to try the downgrade anyway as my next step. So hopefully
that does the trick.
I'll let you know how I make out.
On Jul 14, 2010, at 5:23 PM, Tony Graziano wrote:
> 3 things.
>
> 1. Far en
I already posted this on the forum, but haven't gotten any replies yet. I
know i'm not supposed to send to the list as well, but i'm not sure if it
got sent out. Now that i'm aware of the list i'll use this from now on.
Good evening all,
I am working with sipX 4.2. I set this up about 2 months
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