Check the bridge configuration (domain and proxy setting). The bridge
is sending the INVITE that is suppose to be for the ITSP back to the proxy.
On 03/29/2012 04:03 AM, Tony Graziano wrote:
fyi - one of the polycom firmwares had an issue with a very small
amount of latency and calls to UA's
Hi Douglas,
Yes because call_state_event data have not been imported from redundant server
to primary due to I suppose a problem regarding postgre on the redundant server
Now the problem with postgre is resolved and data are sent correctly to primary
server, but historical data (before I discov
On Wed, Mar 28, 2012 at 7:36 PM, Cyril Constantin
wrote:
> Hi Guys,
>
> Any idea how to force Primary server to collect data from redundant server
> which didn't have been taken from call_state_events table ?
you looking for a manual database import statement?
Hi Guys,
Any idea how to force Primary server to collect data from redundant server
which didn't have been taken from call_state_events table ?
Thanks a lot in advance.
2012/3/27 Cyril Constantin
> Hi Guys,
>
> After a postgresql restart calls made from this server are correctly sent
> to prim
or the phones have leftover outbound stuff or manual config from a past
deployment. Wipe those phones if that is the case.
On Wed, Mar 28, 2012 at 6:17 PM, Nathaniel Watkins <
nwatk...@garrettcounty.org> wrote:
> Agreed – I’m betting on DNS…what is handling DNS for that domain?
>
> ** **
>
>
Agreed - I'm betting on DNS...what is handling DNS for that domain?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Wednesday, March 28, 2012 6:15 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-
If they are both registered, then you surely have a DNS issue or a UA
(phone) issue.
On Wed, Mar 28, 2012 at 5:38 PM, Stiles Watson wrote:
> Numeric only. I'm trying to place a call from ext 145 to ext 141. They
> are on the same subnet and they both got their IPs from the DCHP server
> running
OK. What you have told me makes me ask more questions>
Call Flow:
PSTN number 1234567 calls subsciber 123 at 3335678. UA/subscriber is at IP
x.x.x.x, sipx is at x.x.y.z
Explain what you expected and what happened.
What I am seeing is TCP which usually indicates the UA is a softphone. If
so, WHA
On 3/28/2012 8:39 AM, Douglas Hubler wrote:
> In 4.6 we're using iptables to restrict access to services. This is
> different than 4.4 where we had either clunky, home grown
> authorization schemes (shared secret based) or no protection at all
> (not security risk, just DoS or Buffer overflow vuln
The phone registrations are as follows:
sip:1...@sipx.datatek-net.com
sip:1...@sipx.datatek-net.com
On 03/28/2012 05:38 PM, Stiles Watson wrote:
Numeric only. I'm trying to place a call from ext 145 to ext 141. They
are on the same subnet and they both got their IPs from the DCHP
server run
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <67050>
Message-ID: <105ea.4f738...@forum.sipfoundry.org>
I'm not completely sure what you mean by call flow, but I'll
answer with
Numeric only. I'm trying to place a call from ext 145 to ext 141. They
are on the same subnet and they both got their IPs from the DCHP server
running on the sipX server.
Stiles
On 03/28/2012 05:34 PM, Tony Graziano wrote:
I don't know why anyone would do subscriber lines that weren't numeri
I don't know why anyone would do subscriber lines that weren't numeric
only. Sheesh.
On Mar 28, 2012 5:29 PM, "Nathaniel Watkins"
wrote:
> Are you using numeric or alpha-numeric for your User IDs? Generally,
> numeric is preferred.
>
> --
> This message and any f
Are you using numeric or alpha-numeric for your User IDs? Generally, numeric
is preferred.
This message and any files transmitted with it are intended only for the
individual(s) or entity named. If you are not the intended individual(s) or
entity named you are
fyi - one of the polycom firmwares had an issue with a very small amount of
latency and calls to UA's would fail or fail to media services almost
immediately. In the meantime, in your failed call, explain the call flow so
people can look at it without have to decipher what they think the call was
t
I've updated the logging levels and got a couple new traces.
Attached are the full traces for a failed call.
trace-failed.xml
Description: Binary data
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Thanks. Would be exceptional if it can protect itself out of the box and
set the rules up and then manually administered because then you don't have
to rewrite anything just add the part you need/want.
On Mar 28, 2012 1:28 PM, "Douglas Hubler" wrote:
> re:Switching to manual mode
> no problem.
>
re:Switching to manual mode
no problem.
On Wed, Mar 28, 2012 at 12:11 PM, Matt White wrote:
> I echo what tony said. For example, we change the default ssh port and also
> run agents to monitor server health and replication. Custom ports would be
> nice but as long as we can manually edit iptab
Or if there is a way we can spin up a new box and get the sipx managed
iptables in place and tick a box to tell sipx not to manage them and do it
manually. It would be nice is sipx put the initial rules into place and we
could leave them and tick it off like changing dns to manual...
On Wed, Mar 2
>>> Tony Graziano 03/28/12 11:43 AM >>>
>We use management tools that would require other ports to be open which we
>could do in iptables, so as long as it easy to spin the system up and get the
>default >rules functional we can edit iptables unless sipxconfig will
>overwrite.
I echo what tony
We use management tools that would require other ports to be open which we
could do in iptables, so as long as it easy to spin the system up and get
the default rules functional we can edit iptables unless sipxconfig will
overwrite.
On Mar 28, 2012 11:35 AM, "Douglas Hubler" wrote:
> On Wed, Mar
On Wed, Mar 28, 2012 at 10:27 AM, Tony Graziano
wrote:
> I can see that being able to add a custom rule or two would be nice. i think
> it's great either way though!
We now have an inventory of all the addresses on a system so there
shouldn't be any service/port missing. If there is, development
hurray! (sort of)
at least mystery solved'ish
On Wed, Mar 28, 2012 at 11:16 AM, Aaron Pursell wrote:
> Its a NAT issue, which we should not have but we do. Thanks for the
> replies Tony!
>
> -Aaron
>
>
>
> Aaron Pursell
> Network Systems Administration
> Easter Seals-Goodwill Northern Rocky
Its a NAT issue, which we should not have but we do. Thanks for the
replies Tony!
-Aaron
Aaron Pursell
Network Systems Administration
Easter Seals-Goodwill Northern Rocky Mountain, Inc.
4400 Central Ave
Great Falls, Montana 59405
(406) 771-3721
aar...@esgw.org
>>> "Aaron Pursell" 3/28/201
Jan,
Thanks. May I know what client you are using to test? Is it something
that I can have posession of?
"SipMessage::parseQopValue - no qop value found" is something that I expect.
Response auth hash does not match (bad password?) is the thing that
needs to be looked at
Joegen
On 03/28
hurray!
I can see that being able to add a custom rule or two would be nice. i
think it's great either way though!
On Wed, Mar 28, 2012 at 8:39 AM, Douglas Hubler wrote:
> In 4.6 we're using iptables to restrict access to services. This is
> different than 4.4 where we had either clunky, home
yes DNS is working correctly. No idea at this point, sip to sip
voicemails don't work and pots to sip works.
Aaron Pursell
Network Systems Administration
Easter Seals-Goodwill Northern Rocky Mountain, Inc.
4400 Central Ave
Great Falls, Montana 59405
(406) 771-3721
aar...@esgw.org
>>> Tony Graz
Then the intranet subnets is also showing this as local? a pcap at your
firewall/vpn might shed some light as to why this is doing that. You are
sure the DNS is working properly for "that" vpn segment?
On Wed, Mar 28, 2012 at 9:22 AM, Aaron Pursell wrote:
> Yes I did and no there is only a VP
After you finalized the install, did you send server its profiles?
192.168.1.8:5060 -> 172.16.11.8:5060
Is there a NAT between those?
On Wed, Mar 28, 2012 at 8:48 AM, Aaron Pursell wrote:
> Its 4.4 latest release, installed via repo. If you call via external
> audiocodes fxo it works fine, i
Its 4.4 latest release, installed via repo. If you call via external
audiocodes fxo it works fine, its only sip to sip. I will get around to
a siptrace here sometime this morning.
Aaron Pursell
Network Systems Administration
Easter Seals-Goodwill Northern Rocky Mountain, Inc.
4400 Central Ave
Gr
In 4.6 we're using iptables to restrict access to services. This is
different than 4.4 where we had either clunky, home grown
authorization schemes (shared secret based) or no protection at all
(not security risk, just DoS or Buffer overflow vulnerabilities)
Goals:
- Default rules out of box will
Hi,
can we have the debug level log of the proxy and registrar for this?
joegen
On 03/27/2012 10:34 PM, Jan Fricke wrote:
Hi,
I installed a 4.4 64-bit test system from 287 iso, yum updated it, set
repo to staging and yum updated again.
Sipxproxy is now at release 374.g2acd4. But unfortunately
If it were me (and it's not) I would not be using firmware 3.2.2 on the
phones, rather I would push out bootrom 4.3.1 and firmware 3.2.6 (but
that's me).
(HINT: sipx will do this for you easily).
On Tue, Mar 27, 2012 at 7:02 PM, Tony Graziano wrote:
> You are only showing one side of this. Put
If the calls are through the same itsp then they don't support a hair
pinned call.
On Mar 28, 2012 5:25 AM, "glomos-info" wrote:
> In addition to below.
>
> ** **
>
> We are using version 4.4
>
> The documentation is referring to the "Forward Calls External"
> permission, but where is it
In addition to below.
We are using version 4.4
The documentation is referring to the "Forward Calls External" permission, but
where is it in this version.
Regards, GJ
Van: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] Namens glomos-info
Verzonden: woens
Hi,
When superadmin or a user sets a call forwarding rule (delayed) to an external
phone number, it is not working.
Is there extra configuration required and where? Documentation is poor on this
matter.
Thanks in advance,
Regards GJ
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