Hi:
Our main concern is to know, if Sipx has sip session limitation, users,
conferences, rtp port, etc. With Karoo we have not problem.
Martin Rodriguez
VoIP & Video Engineer
AR: +54-11-4109-1700 ext 8249
US: +1 877 798 8104 ext 8249
UK: +44 20 7043 8269 ext 8249
martin.rodrig...@globant.com
La
If you are using Karoo, then the proxy should have categorized the call
as not natted. If you are running Karoo within a DMZ, make sure you are
using setInterfaceAddress() function in your route.js to set the correct
interface used to connect to sipx. Feel free to send in traces.
On 07/24/20
A graphic won't help anyone troubleshoot it. Post the trace file.
On Jul 23, 2012 8:28 PM, "Kurt Albershardt" wrote:
> Ack - sorry about the large attachment.
>
>
> On Jul 23, 2012, at 18:26 , Kurt Albershardt wrote:
>
> > Getting closer - have AON configured (thanks, Tony) on pfSense and am
> re
Ack - sorry about the large attachment.
On Jul 23, 2012, at 18:26 , Kurt Albershardt wrote:
> Getting closer - have AON configured (thanks, Tony) on pfSense and am
> receiving signaling on 5080. Now sipx is responding to invites with 302
> first and then 407. There's obviously a lot of data
Thanks Tony
I will work with log files and keep downloading nightly
updates. Looking forward to seeing Homer in all its glory.
:d
--
Regards
Mark Dutton
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List Archive: http://list.sipfoundr
naturally i checked all that stuff before emailling the list.
actually it turned out that "special mode" was turned on:
[image: Inline image 1]
why, i don't know, because i sure as shit didn't do it.
any way i can figure out how/when this setting was changed? logs?
On Mon, Jul 23, 2012 at 3:2
Hi Tony,
the server is:
System Information
Manufacturer: Dell Inc.
Product Name: PowerEdge R210
CPU
Intel(R) Xeon(R) CPU X3430 @ 2.40GHz Quad Core
Memory:
free -m
total used free sharedbuffers cached
Mem: 7972 78531
stop panicking. it means in your dial plan.auto attendant someone has
likely changed the default/working/holiday attendants.
Look at the greeting set tp play. it is all there in front of you. If you
have made changes, dont forget to restart services as prompted. make sure
the time/date is correct o
you have not provided much information. How much RAm is being used and how
much is available. With that kind of load I would suspect a resource issue.
On Mon, Jul 23, 2012 at 2:37 PM, Cristian Luna wrote:
> Hi,
>
> I deploy sipx in my environment, with 3000 users in distribuited office
> with DNS
hi all,
my autoattendants are suddenly all giving the message "sorry, our
office is closed" no matter what autoattendant alias/extension you
call (internally or externally). i've actually never even heard this
particular message before (sounds like a sipx default). any ideas for
what i should do
Hi,
I deploy sipx in my environment, with 3000 users in distribuited office
with DNS views and GeoIP, remote users access with karoo, when connect GUI
12 administrator the sipxproxy fail, or when receive 120 calls in
conference in server dedicated and apply a change sipxproxy fail.
Other fail was
sipx-trace is retired... check out homer call trace in the experimental
server options.
On Mon, Jul 23, 2012 at 12:12 PM, Mark Dutton wrote:
>
>
> Hi all
>
> I am trying to find a way to get a sip trace, or live debug
> from the switch.
>
> So far all references to cli tools in the wiki don't se
There are known issues with homer and it is being worked on.
You should still be able to put the logging level at debug for the
component(s) you need. Logs for sipx are (always have been) in
/var/log/sipxpbx
On Jul 23, 2012 12:00 PM, "Mark Dutton" wrote:
>
>
> Hi all
>
> I am trying to find a wa
>
>
>
> Is there any possibility of making this configurable? Specifically,
> synchronizing passwords via check box to synch web portal, voicemail,
> XMPP to a single password?
>
There is one single password for web portal, XMPP, Rest, openACD. Voicemail
PIN is different and we don't have an automa
Hi all
I am trying to find a way to get a sip trace, or live debug
from the switch.
So far all references to cli tools in the wiki don't seem to
exist. I can't find any logs with sip.
I would love a live sip debug like "sip set debug IP xxx" or
"sip set debug peer xx" etc in Asterisk.
I have
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 07/18/2012 12:00 PM, sipx-users-requ...@list.sipfoundry.org wrote:
>>> It's not a bad idea. Many voicemails will have a default of
>>> or or something similar. I have a default that I
>>> apply to all installations when I import my file
Hello,
I dont know if anyone has noticed, but http://wiki.sipfoundry.org/ seems to be
unreachable.
Regards,
Met vriendelijke groet,
Elwin Formsma
Telecats BV
-
Elwin Formsma | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 44 |
Fax: 053 488 99 10 | E-mail:
e.form...@telecats.nl
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