Ok. And calls in both directions work and have audio? I would document that
for the new wiki if it passes all the tests!
I seem to think the same approach would work for voip.ms...
Tony
On Thu, Jul 22, 2010 at 7:54 PM, Michael Scheidell wrote:
> It looks like its sending on port 5070 now . Not
You can definitely edit an existing profile. Maybe the changes weren't sent.
-Original Message-
From: "Michael Scheidell"
Date: Thu, 22 Jul 2010 19:54:31
To: Tony Graziano
Cc: ;
Subject: RE: [sipx-users] if I can get this to work..
It looks like its sending on port 5070 now . Not sure
It looks like its sending on port 5070 now . Not sure why you cent edit an
exiting profile. Inbound works, now level3 needs to verify that they dint need
credentiale.
-Original Message-
From: Tony Graziano
Sent: Thursday, July 22, 2010 7:14 PM
To: Michael Scheidell
Cc: mkitchin.pub.
If these phones are remote or on a vpn that response may never come. if they
are using a firmware version that doesn;t play nice (polycom 3.2.x) it is
just an error message, if the phone is not registered, it will not be ABLE
to respond... provide more info please?
On Thu, Jul 22, 2010 at 7:41 PM,
Can you give a few more details?
Versions for everything involved, phone types, etc.
-Original Message-
From: "McIlvin, Don"
Sender: sipx-users-boun...@list.sipfoundry.org
Date: Thu, 22 Jul 2010 19:41:26
To:
Subject: [sipx-users] Job failure doing restart of phones after send profiles
I have been getting consistent Job Failures on the Restart of phones
initiated from Sipconfig. If I force a manual restart it will reboot and
take on the changes I made.
The Provisioning job is completing OK. But if I elect for an automated
restart - that always fails.
In Job Status it says F
So what does that mean? It works or does not work? If it does not work can
you get a pcap of the failure at your firewall to see what is being
sent/received to determine where it is actually failing?
What "last piece" must be them? Sorry, I'm lost in the translation here.
I actually "dislike" ITS
On 7/22/10 5:37 PM, Tony Graziano wrote:
I think it might be how you are approaching the template.
1. deleted existing profile, restarting services.
2. waiting till they are restarted:
3. devices->gateways
4. add-new gateway->sip trunk
5. name :level3.com'
6. SBC route, sipXb
I think it might be how you are approaching the template.
Use the bandwidth.com template as an example. You have to put in a password
in the template (it does not go to the provider in that example), the
emplate simply requires one.
Tony Graziano, Manager
Telephone: 4
I did both, but, for ip based authentication, .. well, it doesn't
authenticate, but I tried anyway.
had this same problem with voip.ms and static authentication. it just
won't work.
--
Michael Scheidell, CTO
Phone: 561-999-5000, x 1259
> *| *SECNAP Network Security Corporation
* Certifi
Those are 2 different places. One is under configuration (under gateway) and
the other is ITSP account (under gateway)
-Original Message-
From: Michael Scheidell
Sender: sipx-users-boun...@list.sipfoundry.org
Date: Thu, 22 Jul 2010 17:22:12
To:
Subject: Re: [sipx-users] if I can get thi
like I said:
On 7/22/10 5:14 PM, Matthew Kitchin (public/usenet) wrote:
under gateways->itsp config, PORT: i have 5070.
--
Michael Scheidell, CTO
Phone: 561-999-5000, x 1259
> *| *SECNAP Network Security Corporation
* Certified SNORT Integrator
* 2008-9 Hot Company Award Winner, Worl
What about under Gateway, configuration?
On 7/22/2010 4:11 PM, Michael Scheidell wrote:
the 'small guys' seem to work their buns off for .99/month, with 'the
big guys' not caring.
(am I wrong, previously, someone tried to get ATT to work.. same
problem I have/had: we want signaling on udp 5080,
the 'small guys' seem to work their buns off for .99/month, with 'the
big guys' not caring.
(am I wrong, previously, someone tried to get ATT to work.. same problem
I have/had: we want signaling on udp 5080, verizon, same thing)
level3 change the port for me!.
so, I might be able to add to the
>>> Michael Scheidell
NaN. 07/22/10 4:51 PM >>>
>>but when they sent it to port 5060, sipx answered , wiht 200OK, and some
>>stuff.
Sipxbridge does NOT listen on port 5060. It listens on 5080.
5060 is the sip proxy which handles the phones. So if the itsp sends to port
5060 then its handled
I could be wrong, but I would guess that is because if it was on 5060
sipx expected it to be a phone.
On 7/22/2010 3:51 PM, Michael Scheidell wrote:
> On 7/22/10 4:37 PM, M. Ranganathan wrote:
>> On Thu, Jul 22, 2010 at 4:32 PM, Michael Scheidell
>> wrote:
>>
>>> 2010-07-22 16:29:00.359444
On 7/22/10 4:37 PM, M. Ranganathan wrote:
> On Thu, Jul 22, 2010 at 4:32 PM, Michael Scheidell
> wrote:
>
>> 2010-07-22 16:29:00.359444 4.55.00.00 -> 192.168.10.1 SIP Request:
>> OPTIONS sip:204.89.241.150:5080
>> 2010-07-22 16:29:00.364913 192.168.10.1 -> 4.55.00.00 SIP Status: 406
On Thu, Jul 22, 2010 at 4:32 PM, Michael Scheidell wrote:
> 2010-07-22 16:29:00.359444 4.55.00.00 -> 192.168.10.1 SIP Request:
> OPTIONS sip:204.89.241.150:5080
> 2010-07-22 16:29:00.364913 192.168.10.1 -> 4.55.00.00 SIP Status: 406
> Not acceptable
> what do they want? why dosn't sipx lik
2010-07-22 16:29:00.359444 4.55.00.00 -> 192.168.10.1 SIP Request:
OPTIONS sip:204.89.241.150:5080
2010-07-22 16:29:00.364913 192.168.10.1 -> 4.55.00.00 SIP Status:
406 Not acceptable
what do they want? why dosn't sipx like it?
--
Michael Scheidell, CTO
Phone: 561-999-5000, x 1259
> *
Yikes! For both ssh and https, I use completely different background colors
than production just to provide me with one more layer of possible reminder
so I know where I am!
On Thu, Jul 22, 2010 at 2:46 PM, Josh Patten wrote:
> Fortunately it wasn't THAT drastic. Everything was back up in about
1. Because calls to port 5080 do not pass the same rules as a user calling
into the proxy and thereby the DOMAIN name is not checked at all. Port 5080
is used for incoming calls by a trunk provider, so the call is passed if
there is a user part match to send the call to.
2. Don't be fooled by XMPP
On Thu, Jul 22, 2010 at 2:13 PM, Nathaniel Watkins <
nwatk...@garrettcounty.org> wrote:
> Tim – I did this for a while (over a hear) I had sipx on gcgov.local – The
> work around mentioned is under System -> Domain Domain Aliases (see attached
> picture).
Thanks for you response, this has helped
On 7/22/10 2:28 PM, Philippe Laurent wrote:
Been there... before virtualization and bare metal restoration.
A perennial favorite is 'rm -rf /'. Please only use that as a last
resort! :)
Then again, please don't, unless you want to wipe out the server in a
most spectacular way.
I just reboo
On 7/22/2010 1:44 PM, Michael Scheidell wrote:
On 7/22/10 2:34 PM, Matthew Kitchin (public/usenet) wrote:
On 7/22/2010 10:59 AM, Michael Scheidell wrote:
On 7/22/10 11:31 AM, Matthew Kitchin (public/usenet) wrote:
Yes. I'm using Verizon. They say they are using Broadsoft. I
haven;t experience
Fortunately it wasn't THAT drastic. Everything was back up in about 10
minutes but there is nothing like the sinking feeling brought on by
realizing you screwed up the production server.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/22/2010 12:54 PM, To
On 7/22/10 2:34 PM, Matthew Kitchin (public/usenet) wrote:
On 7/22/2010 10:59 AM, Michael Scheidell wrote:
On 7/22/10 11:31 AM, Matthew Kitchin (public/usenet) wrote:
Yes. I'm using Verizon. They say they are using Broadsoft. I haven;t
experienced anything like what you described. The first lev
On 7/22/2010 10:59 AM, Michael Scheidell wrote:
On 7/22/10 11:31 AM, Matthew Kitchin (public/usenet) wrote:
Yes. I'm using Verizon. They say they are using Broadsoft. I haven;t
experienced anything like what you described. The first level techs
were very confused though. They didn't have a clu
Been there... before virtualization and bare metal restoration.
A perennial favorite is 'rm -rf /'. Please only use that as a last resort!
:)
Then again, please don't, unless you want to wipe out the server in a most
spectacular way.
On Thu, Jul 22, 2010 at 2:17 PM, Nathaniel Watkins <
nwatk...@
Did you try ctrl+'Z' :)
Nathaniel Watkins
IT Director
Garrett County Government
203 South 4th Street, Room 210
Oakland, MD 21550
Telephone: 301-334-5001
Fax: 301-334-5021
E-mail: nwatk...@garrettcounty.org
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-user
Tim - I did this for a while (over a hear) I had sipx on gcgov.local - The work
around mentioned is under System -> Domain Domain Aliases (see attached
picture).
You could have the Domain Name as your ".local" and put in an alias that the
outside world would use (i.e. put in sipx.internetdomain
We have all done it
I was in a data center working on a KVM console. I needed to power cycle
the server I was working on. I reached up and hit the power button on
the device right next to the monitor out of habit. I took down a
prominent website for about 2 minutes. Ooops.
On 7/22/2010 12
On Thu, Jul 22, 2010 at 12:58 PM, Tony Graziano <
tgrazi...@myitdepartment.net> wrote:
> You speak "cryptically". I am without my decrypting fluid (coffee) to
> decrypt your statement.
>
I apologize. This seems to be a common problem of mine lately. I'm working
on it. I blame it on the kids for n
On 7/22/2010 11:14 AM, Michael Scheidell wrote:
> On 7/22/10 12:05 PM, Tony Graziano wrote:
>> What port are they sending you calls on? 5080 I hope.
>>
> they seem to think they want to send calls to 5060. I tried to explain
> that this isn't a bunch of phones, that its a call manager.
>
>
Me to
On 7/22/2010 11:03 AM, Michael Scheidell wrote:
On 7/22/10 11:31 AM, Matthew Kitchin (public/usenet) wrote:
Yes. I'm using Verizon. They say they are using Broadsoft. I haven;t
experienced anything like what you described. The first level techs
were very confused though. They didn't have a clu
"... database delete..."
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.myitde
Just in case it happens to someone else: When entering in commands via
SSH on a test sipX server make sure that you are typing those commands
in the test servers SSH session, not the production servers SSH session.
I'll leave it to your imagination to figure out why I'm posting this.
___
You speak "cryptically". I am without my decrypting fluid (coffee) to
decrypt your statement.
Firewall translation is one thing, those are correct statements.
Your ITSP should send calls to port 5080 (not 5060). The proxy runs on port
5060 and remote users connect via port 5060.
Noone seems to k
There is a separation of ports using sipxbridge:
port 5060 - remote users
port 5080 - trunks
Depending on your carrier, they will send the calls on whatever port you
register on (ex: voip.ms). With ITSp's that use IP based ACL, you configure
your account for them to send the calls on a specific p
Thanks for your response, Nathaniel. This makes sense.
At this point I'd rather not change the sipXecs machine name, which would be
required if I used a solution similar to yours.
If possible and not too complicated, I'd like to use the workaround
mentioned (but not described) in the "DNS Concept
On 7/22/10 12:24 PM, Josh Patten wrote:
Unless you use a separate SBC, yes 5080 is a requirement.
the ITSP will never send me anything on 5060?
maybe I so something perverted like in/out port mapping? or just give up
and put freeswitch in between, or opensbc?
(didn't someone here have the s
5080 is a sipx requirement for trunking.
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:
h
Unless you use a separate SBC, yes 5080 is a requirement.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/22/2010 11:23 AM, Michael Scheidell wrote:
On 7/22/10 12:20 PM, Tony Graziano wrote:
Tell them you need the calls sent to 5080. That's why your in
On 7/22/10 12:20 PM, Tony Graziano wrote:
Tell them you need the calls sent to 5080. That's why your inbound
calls are having audio issues, the media relay is kicking in likes its
a remote user.
I don't understand their logic in using an IP ACL and also requiring
authentication on outbound
Tell them you need the calls sent to 5080. That's why your inbound calls are
having audio issues, the media relay is kicking in likes its a remote user.
I don't understand their logic in using an IP ACL and also requiring
authentication on outbound calls.
On Thu, Jul 22, 2010 at 12:14 PM, Micha
On 7/22/10 12:05 PM, Tony Graziano wrote:
> What port are they sending you calls on? 5080 I hope.
>
they seem to think they want to send calls to 5060. I tried to explain
that this isn't a bunch of phones, that its a call manager.
>
> Tony Graziano, Manager
> T
What port are they sending you calls on? 5080 I hope.
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
On 7/22/10 11:31 AM, Matthew Kitchin (public/usenet) wrote:
Yes. I'm using Verizon. They say they are using Broadsoft. I haven;t
experienced anything like what you described. The first level techs
were very confused though. They didn't have a clue.
We don't authenticate at all. It is all done by
I don't see the point of using multiple authentication mechanisms.
Its being offered and so is G711. If they won't honor G722 why shouldn't
they also see G711 and honor that?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepar
On 7/22/10 11:31 AM, Matthew Kitchin (public/usenet) wrote:
Yes. I'm using Verizon. They say they are using Broadsoft. I haven;t
experienced anything like what you described. The first level techs
were very confused though. They didn't have a clue.
We don't authenticate at all. It is all done by
Yes. I'm using Verizon. They say they are using Broadsoft. I haven;t
experienced anything like what you described. The first level techs were
very confused though. They didn't have a clue.
We don't authenticate at all. It is all done by IP. I had to tell them I
was running Nortel SCS.
On 7/22
your ITSP using broadsoft?
or, you have broadsoft sip trunks and are routing to/from sipx?
(any broadsoft engineers want to ping me so I can tell the level3 guys
how to set it up?)
they want me to authenticate on every outbound call. sipx thinks a hold
is a outbound call (sorta?) a reinvite
on phone with level3. great sip prices, but they are using a broadsoft
call manager to provide me ip trunks.
strange setup:
they want me to use ip based REGISTRATION, but on a per call basis, they
need me to use the username/password authentication.
I would think this is REALLY terrible, as
On Wed, Jul 21, 2010 at 9:49 PM, Douglas Hubler wrote:
> please reply to me off list. I only need one volunteer.
I'm all set on this, thanks everyone
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/ar
A 200 OK to an INVITE must always be ACKed. Feel free to send a
snapshot so I can double check it. I will not be looking at 4.0.4
traces.
Thanks
Ranga
On Thu, Jul 22, 2010 at 8:52 AM, Sven Evensen wrote:
> We have a call from a mobile A coming in through the SIP trunk to our IP
> PBX, this ca
Shall I enter a JIRA ticket?
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/21/2010 8:55 PM, Martin Steinmann wrote:
> Josh
>
> There certainly could and I see no reason why that couldn't be done. It's
> not the first time this gets suggested
> --martin
>
This is a subdomain, but also the hostname.
Andreas
On 22/07/10 12:47, Tony Graziano wrote:
> Is that your domain or hostname? If it is your domain it would be a
> subdomain, correct?
>
>
>
> Tony Graziano, Manager
> Telephone: 434.984.8430
> Fax: 434.984.8431
>
>
Is that your domain or hostname? If it is your domain it would be a
subdomain, correct?
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
I actually noticed this one in freeswitch.log:
2010-07-22 12:23:19.840174 [WARNING] switch_core.c:1046 Cannot locate
domain voip.wri-irg.org
Can this be related to the problem?
Andreas
On 22/07/10 08:26, Andreas Speck at WRI wrote:
> I have to let this rest a bit, as I have other urgent things
I have to let this rest a bit, as I have other urgent things to do
now, so will need to get back to it once I have time again (will at
least be two weeks).
What I forgot to mention is that Voicemail also doesn't kick in when
calls come in via PSTN, sip trunk, or Skype gateway and don't get
ans
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