On 8/7/10 8:18 PM, Rolland Hart wrote:
call on hold and 7sec after being placed on hold the call drops.
is this something NEW, or your first install?
what to check: pfsense. make sure you follow tony's guide and make sure
you have static port nats.
is this a new ITSP? does it work on any oth
Do not know if this issue has been discussed before and if it has I would
sure love the link to the discussion.
The problem I am having is, I place a call to an outside number from inside
my network. The call is answered on the other end. i place the call on hold
and 7sec after being placed on hol
On 8/7/10 6:13 PM, Tony Graziano wrote:
That seems actually hard to implement because it adds a layer to all
calls at the proxy. Have you looked at the sipvicious scripts yet that
allow it to be recognized and killed?
pfsense with the snort package could deal with these.
there is also a 'kill
I would try send profiles from the services screen. You can also restart
FreeSWITCH (media services) from the service screen.
mp3: I don't see a problem with licenses using an external mp3 encoder that
you can install optionally. We just need to find someone who can provide a
patch. We woul
so, issue #2: useing mod_shout to send mp3 voicemails won't work either.
I think I will work up a script to do this then.
yuck. I did something BAD. freeswitch won't come up now.
so, how do I reload freeswitch?
On 8/7/10 6:26 PM, Martin Steinmann wrote:
sipXecs does not use the FreeSWITCH
Mike
sipXecs does not use the FreeSWITCH voicemail system. We only use
FreeSWITCH as the media server, but built our own VM application.
I think it is hardcoded for now. You can do "grep postmaster *" in the
source directory sipXivr/src/main/java/org/sipfoundry/voicemail
We should pr
On 8/7/10 6:18 PM, Michael Scheidell wrote:
On 8/7/10 6:15 PM, Michael Scheidell wrote:
On 8/7/10 6:11 PM, Michael Scheidell wrote:
what do I need to run to get freeswitch to restart?
rebooted the server, didn't help.
So, where are these settings REALLY.
found /var/log/sipxpbx/freeswitch.xm
On 8/7/10 6:15 PM, Michael Scheidell wrote:
On 8/7/10 6:11 PM, Michael Scheidell wrote:
what do I need to run to get freeswitch to restart?
rebooted the server, didn't help.
So, where are these settings REALLY.
found /var/log/sipxpbx/freeswitch.xml.fsxml
assuming that is loaded from database
On 8/7/10 6:11 PM, Michael Scheidell wrote:
what do I need to run to get freeswitch to restart?
rebooted the server, didn't help.
So, where are these settings REALLY.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
> *| *SECNAP Network Security Corporation
* Cert
I've thought about it, but they also support remote workers on home cable
modems.
I've been thinking of moving them to vpn's.
-M
>>> "Todd Hodgen" 08/07/10 5:49 PM >>>
That seems actually hard to implement because it adds a layer to all calls
at the proxy. Have you looked at the sipvicious scripts yet that allow it to
be recognized and killed?
On Sat, Aug 7, 2010 at 5:42 PM, Matt White wrote:
> Yes, that is exactly the scenario I'm describing.
>
> This custome
many servers who have overly aggressive anti-spam systems will try to
block backscatter by blocking emails from 'postmaster'. (dumb thing to
do, lots of auto responders use postmaster@)
Sipx also uses 'postmas...@localhost.{domain}' in the From line
(I used domain masqurading to get rid of the
How about a temporary fix by blocking all 5060 traffic that does not come
from your own firewall list - ITSP, Support IP addresses, Remote locations,
etc., and then blocking all others? From a security standpoint, it's
probably the right thing to do.
From: Matt White [mailto:mwh...@thesummit-g
Yes, that is exactly the scenario I'm describing.
This customer actually already has a call block feature with their ITSP...ie to
block anonymous calls and a few others. But the calls did not cease. When we
looked into it the calls where not coming in via the SIP trunk but directly to
port 5
Congratulations Martin, Jerry, Mike, Doug and the rest of the team. It's an
exciting day for all of you!
The site looks great, love the graphics! Best of luck to eZuce.
From: sipx-dev-boun...@list.sipfoundry.org
[mailto:sipx-dev-boun...@list.sipfoundry.org] On Behalf Of Martin Steinmann
S
Btw - nice analogy.
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 Contract Customers:
http://www.myitdepartm
Hence what he is asking for is a call screening feature. Turn it on for a
user, media server asks you to say your name, then rings you and plays the
audio "who it is".
This is not an authorization thing, rather its a call screening feature. I
imagine it would have to be inmplemented in the proxy a
There is an analogy that works well here. Today, you can call any telephone
number you want, ring the phone and hang up. This isn't much different, a
user can use sip to call directly into a sip phone. And, as kids I think
many of us can recall playing pranks on people over the phone - caller I
But what you describe is a user privacy feature. Internal calls provide full
username/line. The callee can hit dnd, reject or ignore the call. That is
internal to the proxy. If I am called from a site and a number in their
huntgroup shows up instead of their mail pilot number, how will I get the
in
Right, which is why I wondered why more ITSP that send on port 5060 do not work
for inbound calls.
As a test, I just set my test vitelity.net account to not use registration and
forced it to use port 5060. Test call comes in and works has audio.
Transfer fails...which I assume is because they
I don't think any RFC changes are required at all. Invites are challenged all
the time.
Example,
Ext 345 sends invite to sipxproxy for ...@sip.com
sipxproxy responds "404 not authorized"
Ext 345 resends invite with authorization header
sipxproxy checks authorized.xml and process the call
Sipxrelay anchors the media. Sipxbridge separates trunking from pure
sip and remote users and acts as a gateway out via carrier if the
proxy authorizes the call. Sipxproxy sends the call (or rejects it) if
there is validity to the invite to the user.
On 8/7/10, Matt White wrote:
> Yes, I could
Then you would have to invent an authorization rfc for an simple invite,
which kind of breaks the intent of sip in a way. Invites from the internet
to the proxy (port 5060) can only reach your system (AA, conferernce, media,
users), not place calls.
Itsp's require auth to send calls to them. Sipx
Yes, I could try and get the ip range from the itsp but I know they use
re-invited to select different upstream providers.
It would seem there should be a way to require authentication for the inbound
call. I dial plan in reverse.
I'm not saying to require authorization before the invite, I'
Yeah, I knew that it would for remote workers by determine the SIPX-NONAT
status , I guess I didn't realize it would for non registered sip UA like an
inbound sip call.
Makes me wonder how hard it would be to get SipXproxy to work with ITSP that
must send to port 5060 instead of port 5080. If
The proxy handles NAT and anchors media. This is used for remote workers
and happens automatically
--martin
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matt White
Sent: Saturday, August 07, 2010 9:53 AM
Cc: sipx-users@list.sipfound
Yes, I was just surprised that the call had two way audio considering the call
was traversing a NAT and not going through sipxbridge which anchors the media.
-M
>>> Tony Graziano 08/07/10 8:46 AM >>>
Calls to the system will have a dest port of 5060 (sip), not 5080, they are
not itsp calls go
We get the same behaviour. It seems to happen any time we do anything with CDR
records. We run very successfully under a VMWare environment.
Multiple systems,HA etc.
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Calls to the system will have a dest port of 5060 (sip), not 5080, they are
not itsp calls going through sipxbridge. This is how one sip system can call
another.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/T
No. It's a sip call. That's perfectly normal. ANY sip call from the Internet
to a user on the system would not be requiring permissions or even to pass
through a gateway. Only outgoing calls do that.
Probably a sipvicious variant. If your system is setup properly I'm not sure
I would worry. If you
On Thu, Aug 5, 2010 at 9:55 AM, Geoff Van Brunt wrote:
> I do however agree with you about the problems with that approach. One
> comment I would like to make in general about the project; there is no
> formal process for "who is responsible for what".
We're still reorganizing, but I agree. I go
I have a client that looks like its getting calls from Russia on port 5060. It
is not coming from the bridge.
The calls come in from various sources and seem to be sending invites to random
common extensions causing the phones to ring.
I would have expected the invite to be challenged as this
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