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I think its fixed in the 4.6 in the staging area but 4.6 needs some cleanup
still...
On Jul 7, 2012 6:23 PM, Kurt Albershardt k...@nv.net wrote:
Is there a dependency on mysql that's not being met by 'yum groupinstall
sipxecs' ?
The only mysql package I show is:
.
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chalk that us to lack of coffee. I forgot I had been.
On Fri, Jul 6, 2012 at 8:43 AM, Douglas Hubler dhub...@ezuce.com wrote:
you should have been prompted during the install to enter your own root
password
On Fri, Jul 6, 2012 at 8:24 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote
this
manually before the sipx script runs?
FWIW - To be more easily usable to some of us, it would be nice to include
both nano and wget packages during the ISO install like 4.4 did.
On Fri, Jul 6, 2012 at 8:53 AM, Tony Graziano
tgrazi...@myitdepartment.netwrote:
chalk that us to lack of coffee. I
thanks!
On Fri, Jul 6, 2012 at 10:24 AM, Laurentiu Ceausescu lauren...@ezuce.comwrote:
On Fri, Jul 6, 2012 at 4:52 PM, Laurentiu Ceausescu
lauren...@ezuce.comwrote:
On Fri, Jul 6, 2012 at 4:45 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
(using the last ISO in the sipx-stage
into the system.
** **
*From:* sipx-dev-boun...@list.sipfoundry.org [mailto:
sipx-dev-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
*Sent:* Friday, July 06, 2012 5:24 AM
*To:* sipXecs developer discussions
*Subject:* Re: [sipx-dev] Install of 4.6 ISO in State Area notes
to
generate ssl keys and a script to install them again (it's not
/usr/bin/ssl-cert/gen-ssl-keys.sh)
Someone clue me in? Also, what are the default credentials for Homer?
On Fri, Jul 6, 2012 at 2:03 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Fri, Jul 6, 2012 at 10:39 AM, Tony Graziano
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Also, are you sure the use local ip is set up in eyebeam and NAT/ICE/TURN
are deselected and your intranet subnets reflect the fact you are remote?
On Wed, Jul 4, 2012 at 12:28 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
I was drawn to the 406 and followed it to the codecs. While I
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http
of a logical error?
Sven
On Wed, Jul 4, 2012 at 5:48 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Well, I have to ask an obscure question... are your virtual kernels up to
date and not having any leap second issues.
It all boils down to timing (I think). I have a suspicion
it a few days.But it worked yesterday!
Sven
On Wed, Jul 4, 2012 at 5:31 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Also, are you sure the use local ip is set up in eyebeam and
NAT/ICE/TURN are deselected and your intranet subnets reflect the fact you
are remote?
On Wed, Jul 4, 2012
I know where you are coming from ranga. Still nice to hear from you.
I don't think anyone here has much mire than a passive interest in what
avaya is doing. I certainly wouldn't ask you about avaya or to break any
confidences.
I was hoping that if we make some changes to sipxbridge code we might
/
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I'll ditto the price preference on that call volume. We would prefer not to
deal with trunk or internet quality if we don't have to.
On Jul 3, 2012 12:27 PM, Matt White mwh...@thesummit-grp.com wrote:
Sipxbridge is more than capable of handling 24 simultaneous calls through
a siptrunk if the
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I checked now and the email below are in whitelist now . As it seems the
email/domain reported
using assp-white@ are not added instantly to the whitelist file .
Please after which event or time the email/domain are added to whitelist file ?
Thank you
Graziano
Hello
I am trying to add
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(also, if you are local to the server or not registering via sipxrelay, why
is stun enabled on bria?)
On Mon, Jul 2, 2012 at 9:38 AM, Tony Graziano
tgrazi...@myitdepartment.netwrote:
This shows it successfully registered BUT also shows a problem making a
connection on port 5061 after
Graziano
tgrazi...@myitdepartment.netwrote:
(also, if you are local to the server or not registering via sipxrelay,
why is stun enabled on bria?)
On Mon, Jul 2, 2012 at 9:38 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
This shows it successfully registered BUT also shows a problem
to users list w/appropriate caveats.
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This like comes up in wiki search
http://wiki.sipfoundry.org/display/sipXecs/List+of+Features
but it does not work.
I am not sure how a link breaks with atlassian but am guessing its an
indexing issue?
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and Tony for your help as usual.
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/archive/sipx-users/
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an
example of how you would emulate the E.164 inside a dial plan?
Thanks,
Matt
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
[tgrazi...@myitdepartment.net]
Sent: Wednesday, June 27, 2012 7:59 AM
Depending on the phones you pick, you can do this from device groups too.
On Jun 28, 2012 4:14 PM, Michael Grosh michael.gr...@patlive.com wrote:
I am setting up our phones to pick their ntp server from a
dchp server located locally.
by default all our phones were using pool.ntp.org. I
*@adlinkd.it *@adlinke.net;
Jun-28-12 08:42:50 Email: whitelist addition: *@adlinkd.it;
Jun-28-12 08:42:50 Email: whitelist addition: *@adlinke.net;
If I check the /usr/local/assp/whitelist file there is not trace of
*@adlinkd.it and *@adlinke.net
Thank you
Graziano
You would do this in the dial plan. I've always been able to send a + using
a basic dial plan entry.
On Jun 27, 2012 7:06 AM, m...@mattkeys.net m...@mattkeys.net wrote:
I can't seem to get sipxecs to pass along a + as a ENUM prefix. I've
entered e164.org in the base domain, and + in the add
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On Sat, 23 Jun 2012 16:36:38 -0400, Tony Graziano wrote:
That will work but not the best way to do it (MWI may have an
issue,
etc.).
Is the customer auto provisioned (devices) via sipx?
On Sat, Jun 23, 2012 at 3:00 PM, m...@grounded.net
m...@grounded.net
wrote:
On Sat, 23 Jun
://list.sipfoundry.org/archive/sipx-users/
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On Mon, Jun 25, 2012 at 1:05 PM, Michael Picher mpic...@ezuce.com wrote:
Certificate?
try re-generating your self signed certs...
On Mon, Jun 25, 2012 at 12:41 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I think you really need to :
1. Upgrade to 4.4
2. Get a snapshot
configuration is it still a good way to link to the outside world?
thanks~
On Jun 23, 2012, at 10:59 , Tony Graziano wrote:
Therein lies your problem. At no time has multiple NIC's ever been
supported or recommended.
I would suggest you drop back down to one NIC. The issue you are having
sipXbridge a whirl.
I notice that http://wiki.sipfoundry.org/display/sipXecs/SIP+Trunking has
not been updated since April 2011. Are there changes I should be aware of?
On Jun 24, 2012, at 9:28 , Tony Graziano wrote:
Sipxbridge is just an option. A properly configured SBC works well too
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Graziano
*Sent:* Saturday, June 23, 2012 12:59 PM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout
** **
Therein lies your problem. At no time has multiple NIC's ever been
supported or recommended.
** **
I
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if something more is necessary.
Thanks, I will take care of.
George
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On Jun 17, 2012, at 13:10 , Tony Graziano wrote:
(this is also a way of saying pfsense cannot handle the record type sipx
needs, it needs a real dns server to be authoritative for it, not just an A
record)
On Sun, Jun 17, 2012 at 3:09 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote
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http
if the prerequisites for mongo and other new packages can be met easily
before anyone would think about pursuing it)
On Mon, Jun 18, 2012 at 3:16 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
That would also require a driver for the OS and I think you would have
better luck developing
...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
*Sent:* Saturday, June 16, 2012 5:51 PM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout
** **
its a grandstream
,
SIPStatus=Running,
sipXrecording=Running,
CallResolver=Running,
SipXrest=Running,
SIPXProxy=Running}
** **
*From:* sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
*Sent:* Sunday, June 17, 2012 6:12 AM
}
** **
** **
*From:* sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
*Sent:* Sunday, June 17, 2012 9:30 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout
Not exactly a straight answer... Pfsense can hand out the dhcp options the
same as sipx. I would consider DNS forwarding to the sup domain and do this
quite regularly. This way sipx is responsible only for its sip domain /DNS.
But that's me. I typically deploy sipx as a sudimain and keep things
Not exactly a straight answer... Pfsense can hand out the dhcp options the
same as sipx. I would consider DNS forwarding to the sup domain and do this
quite regularly. This way sipx is responsible only for its sip domain /DNS.
How I do it -- I typically deploy sipx as a subdomain and keep things
(this is also a way of saying pfsense cannot handle the record type sipx
needs, it needs a real dns server to be authoritative for it, not just an A
record)
On Sun, Jun 17, 2012 at 3:09 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
Not exactly a straight answer... Pfsense can hand out
Did you try karoo?
On Jun 16, 2012 8:17 AM, Gerry Hull ge...@telosity.com wrote:
Hi Joegen,
Sorry for the OT. Hey, you wrote a useful piece of code... so why not use
it? I use sipXecs for other projects, but in this case I just need an SBC.
I'll shutup now.
Gerry
On Fri, Jun 15, 2012
to 4.6 from remote.
Mike
Thanks,
Mike
On Fri, Jun 15, 2012 at 4:10 PM, m...@grounded.net m...@grounded.net
wrote:
I was... afraid of that... but appreciate the reply. Thanks very much.
On Fri, 15 Jun 2012 16:03:44 -0400, Tony Graziano wrote:
For sipx 4.4, 5.8 is used. I am
:* sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
*Sent:* Saturday, June 16, 2012 11:24 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout
** **
You
The quick fix would seem to be scheduling a cron job that sends a query like
this to the InterMapper database:
DELETE FROM event WHERE end_time now() - '30 days'::interval CASCADE;
(adjusting the interval as appropriate - the formatting is very friendly and
described in the Postgres
We use Patton fxs gateways.
Faxing is a crap shoot. What is your PST connectivity? We find unless you
are using pots lines, pri or sip trunk with t.38, it will not be worth the
effort.
On Jun 15, 2012 5:02 PM, gabriel g...@bayintegrated.net wrote:
hello sipx-users,
I am running a brand new
from the grandstream and
the likes...
-gabriel
On Fri, 15 Jun 2012 17:09:22 -0400, Tony Graziano
tgrazi...@myitdepartment.net wrote:
We use Patton fxs gateways.
Faxing is a crap shoot. What is your PST connectivity? We find unless
you
are using pots lines, pri or sip trunk with t.38
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Use a better phone. The m3 is a consumer grade device.
I have no issues with timing on 4960's when configured properly.
On Jun 12, 2012 2:44 AM, milosz mew...@gmail.com wrote:
It is probably not getting or missing an ack to indicate the call
established. This would indicate a carrier or
Years ago they worked fine. The firmware updates killed it.
Wth are you doing g711 pass through for faxing? The only reliable way to
FAX via sip is using t.38. We do this with the 4960 and smart nodes/fxs
ports without issue. Its 2012, not 2002, g711 is a crappy crap shoot.
On Jun 12, 2012 11:02
/
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M3's are not even made by snom. They are oem'd consumer phones with the
snom name on them. In Europe you may see the same phone but with different
names on them.
It is probably not getting or missing an ack to indicate the call
established. This would indicate a carrier or gateway config issue.
on it.
Will it be possible for example to backup 4.4.0 to restore it onto the new
version coming?
Mike
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Yes. This would really depend on the configuration of any firewall in
between though.
On Jun 7, 2012 8:28 PM, Noah Mehl n...@tritonlimited.com wrote:
It seems that sipxecs is ignoring my sip registrations from soft phones
from the world. Is sipxecs configured to not respond to 5060 from the
, respond to
register from anywhere?
~Noah
On Jun 7, 2012, at 10:17 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
Yes. This would really depend on the configuration of any firewall in
between though.
On Jun 7, 2012 8:28 PM, Noah Mehl n...@tritonlimited.com wrote:
It seems that sipxecs
Or any Patton smart node.
On Jun 4, 2012 10:37 PM, Jimmy dimitri_mano...@yahoo.com wrote:
Guys what do you suggest and works remotely for an analog
ATA?
Need to do some calls with an analog phone. What plays well
with NAT?
MP-114 FXS or SPA2102?
Thanks in advance
Are you using anything to connect to postgres remotely to collect cdr
records? Are there any users logged in tying to run cdr reports or view Dr
records?
If you restart service for cdr (call resolver) does the message continue?
On Jun 5, 2012 5:23 AM, Henry Dogger h.dog...@telecats.nl wrote:
Government,
203 South Fourth Street, Courthouse, Oakland, Maryland 21550
www.garrettcounty.org
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ERROR: duplicate key violates unique constraint cdrs_call_id_unique
** **
It is completely filled with these lines…
** **
*From:* sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
*Sent:* dinsdag 5 juni 2012 11:49
** **
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...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
*Sent:* dinsdag 5 juni 2012 14:16
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] loss of connection to database
** **
Wow. Sounds like there is a problem
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This usually means your log level is above notice.
On Jun 4, 2012 3:21 PM, Saad skhan...@gmail.com wrote:
Hello,
We are seeing tons of the following SIP messages in our sipXproxy logs and
want to stop them as it seems to affect the system functionality ( call
dropping, call setup .. etc) ..
I think the port is now 9095
On Jun 3, 2012 6:52 PM, Noah Mehl n...@tritonlimited.com wrote:
My first problem is that my open fire is not using the SSL Cert I uploaded
using the SipX web control panel. So I tried to enable the open fire
console with these directions:
%25252525252525252525252525253Furl%25252525252525252525252525253D%2525252525252525252525252525252Findex.jsp
Same thing :( Does the host name matter?
~Noah
On Jun 3, 2012, at 7:35 PM, Tony Graziano wrote:
I think the port is now 9095
On Jun 3, 2012 6:52 PM, Noah Mehl n...@tritonlimited.com
.
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Fax: 434.465.6833
I am seeing this after trying to connect with 3cx, csipsimple, etc. and
found it suddenly starts working if I switch the remote phone config to TCP
instead of UDP.
Before I open a JIRA is anyone else seeing this?
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi
Siopx does not support multiple interfaces. Is this what you are trying to
do?
On May 31, 2012 8:22 AM, Daniel Peinado Lopez daniel.pein...@iant.de
wrote:
Hello,
I'm configuring Audiocodes Mediant 1000-MSBG (Firmware 6.00AL.018.003)
with SipXecs - 4.4.0 - centos.
First I have installed the
Managing the gateway by Sipx is optional. Either Sipx can
manage it or you can manage it yourself directly.
If for some reason sipx configured the web interface off by turning it off
in the config file you can verify this by looking at the config file itself
(download via sipxconfig). You can
Did you check your arp cache? Did you check your arp AFTER trying to ping
it? Can you ping the m1000 from with SSH in sup?
On May 31, 2012 9:26 AM, Daniel Peinado Lopez daniel.pein...@iant.de
wrote:
Hello,
maybe the problem is that SipXecs is installed in a module inside the
AudioCodes M 1000
/archive/sipx-users/
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~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
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Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
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by the firewall or NAT.
Ray
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~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
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~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony
-5001
Fax: 301-334-5021
E-mail: nwatk...@garrettcounty.org
** **
*From:* sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
*Sent:* Thursday, May 31, 2012 1:24 PM
*To:* Discussion list for users of sipXecs software
to my Incoming
problem.
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Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi
Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
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LAN/Telephony/Security and Control Systems Helpdesk
.
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Tony Graziano, Manager
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Fax: 434.465.6833
know :)
~Noah
On May 30, 2012, at 2:00 PM, Tony Graziano wrote:
did you send the server its profiles and restart services?
typically I do this and also if its a major update (4.2 -to 4.4) I also
reboot.
On Wed, May 30, 2012 at 12:51 PM, Noah Mehl n...@tritonlimited.comwrote:
I updated
Gasp. You are right AND I instigated that!
On Wed, May 30, 2012 at 5:57 PM, George Niculae geo...@ezuce.com wrote:
On Thu, May 31, 2012 at 12:53 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Usually the server will send profiles do all of the device if upgrade is
done.
Actually
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 |
Op 21 mei 2012, om 16:13 heeft Tony Graziano het volgende geschreven:
Perhaps: misspoke
...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, April 20, 2012 12:10 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Fax Problems (Big Time)
did you do that? if so, did it fix it?
On Fri, Apr 20, 2012 at 11:53 AM, Robert Schroeder
robert.schroe
http://track.sipfoundry.org/browse/XX-8904
On May 28, 2012 12:38 PM, Saad skhan...@gmail.com wrote:
Yes, Thank you :) .. from the SIpxconfig file it looksmlike the FTP server
is replying with error 421 and closing connection after .. Sound s like FTP
server issue ..
Regards
Saad
On Mon,
Because you are thinking you can't keep a hot unconfigured sipx VM spun
down to use at any time?
Build you VM from ISO but don't run the sipx setup script. Spun it up to
update as needed via yum. When you need it spin it up and restore it. This
method doesn't care if it is bare metal or virtual
That is correct. We use appia but you need a minimum of 250.00 billing per
month to setup an account t with them and be clear you need t.38 enabled
trunks. You have to set your re-register intervals to very short periods
(180 seconds), otherwise its normal.
On May 23, 2012 7:59 AM, Mike Pinkerton
, but only occasional -- perhaps less than 100 per month.
Any suggestion on a trunk provider for a low volume use case?
Thanks.
--
Mike Pinkerton
On 23 May 2012, at 08:12, Tony Graziano wrote:
That is correct. We use appia but you need a minimum of 250.00 billing per
month to setup
Is this a virtual? How much ram? The price sounds too low to be bare metal
so there are scalability issues with virtual stuff in general (media)
depending on the platform used.
On May 23, 2012 11:17 AM, m...@grounded.net m...@grounded.net wrote:
I haven't looked anywhere other than appia so far
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