4)
> >> at java.lang.Thread.run(Thread.java:679)
> >>
> >
> > Looks like the issue reported by Tony today. Let me reinstall a fresh
> > system and get back to you
> >
>
> Hm, I cannot reproduce this problem, are you testing 32 or
t actually sounds like what I'm looking for, I've tried checking the
> wiki, but I can't locate it. Is there a list of third-party
> plugins/software out there? Apologies for all the noob queries.
>
> Aaron
>
> On Thu, Oct 4, 2012 at 11:52 AM, Tony Graziano <
*
>
> InSysPro, LLC
>
> M. 954-557-2771
>
> *ja...@insyspro.com*
>
>
>
>
>
>
>
> ___
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> sipx-users@list.sipfoundry.org
> List Archive: http://list
e: http://list.sipfoundry.org/archive/sipx-users/
>
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see here:
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On Thu, Oct 4, 2012 at 10:40 AM, Tony Graziano wrote:
> attached
>
> On Thu, Oct 4, 2012 at 9:37 AM, George Niculae wrote:
>
>>
>>
>> On Thu, Oct 4, 2012 at 4:35
e changes using udev
rules file to match a MAC address with a name, but i don't known how to
make a similar operation on a leaf linux system
Thanks
Graziano
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e
>
> On 10/04/2012 07:21 PM, Joegen Baclor wrote:
>
> If you sanitized the logs, you pasted, then that should be ok. No caller
> alias configured for identity=so_and_so is actually not an error. We must
> get rid of this and promote to debug
>
> On 10/04/2012
The identity was 200@sipdomain
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st.EslRequestScopeRunnable.run(EslRequestScopeRunnable.java:24)
at java.lang.Thread.run(Thread.java:679)
3. Cannot call IVR or AA
User can call its own extension which instead go to VM and generate #2
above.
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Tom
it seems to be a routing problem: which is the default route ip of your
router?
I think it is your ISP... A solution could be to use an SSH tunnel from
your client to your astlinux system and use it to connect to your router
bye
graziano
PS: if you want to use astlinux as vpn
> >>>> Regards,
> >>>> Kumaran T
> >>>>
> >>>> ___
> >>>> sipx-dev mailing list
> >>>> sipx-dev@list.sipfoundry.org
> >>>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
> >>>>
> >&
Make sure server behind nat is disabled.
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perhaps can be a good solution use a 2,5" Solid State Disk or a 2,5" 44
pin IDE hard disk in behalf of compact flash, or use an external USB HD
to store log files...
I remember to have an old astlinux (0.4.x) installation on a net4801
using a 2,5" IDE 44 pin HD...
bye
Graziano
cation to the
> network admin the IP has been uploaded to the central server. In other
> words you can play offense as well as defense, if you so choose.
>
> ** **
>
> ** **
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of
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Using or developing for sipXecs fro
Right. Unless I start supervisor, proxy and registrar with nofork I can't
register a phone. Will there be a fix based in the JIRA already mentioned
or do I need to provide more information?
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the service(s) will start using nofork and show running in sipxconfig. If I
start supervisor, proxy and registrar using nofork, I can register the
phone too.
On Sun, Sep 30, 2012 at 10:00 AM, Douglas Hubler wrote:
> On Sun, Sep 30, 2012 at 8:49 AM, Tony Graziano <
> tgrazi...@myitdepar
yes, testing is 32 bit.
On Sun, Sep 30, 2012 at 9:28 AM, George Niculae wrote:
> On Sun, Sep 30, 2012 at 4:12 PM, Tony Graziano
> wrote:
> > It says it is running. I can't register a phone. I've shut down the
> test. Is
> > this fix built yet?
> >
>
>
It says it is running. I can't register a phone. I've shut down the test.
Is this fix built yet?
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here is no log), and cf-agent log for system is
blank. I don;t think it should be empty. Yet there is nothing wrong with
my install process that should cause this, is there? sipxsupvisor status
says cf-serverd is running. If I restart it I get service RUNNING but I
should not have to do this (fr
He would need to have access to the registrar rpm which is not made
available anymore via the download site.
This would not be a certificate issue. If it was a certificate issue the
only noticeable problem would be people cannot leave voicemail.
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I did the step to ensure the db files were in mongod and chown'ed them. I
am not able to get mongo to run. I am not able to get into sipxconfig.
Is it just me?
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I think it can (inbound). Outbound with sipxbridge would in turn introduce
asynchronous routing but inbound via IP will work.
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It finally started working on a Polycom. But I do not get to
redirected IVR when noone answers. So I guess i am looking at an
update and trying again. I still cannot register with a softphone.
On Wed, Sep 26, 2012 at 11:36 AM, Gerald Drouillard
wrote:
> On 9/26/2012 11:14 AM, Tony Graziano wr
n this
case.
On Wed, Sep 26, 2012 at 11:10 AM, Philippe Laurent wrote:
> No, but I've not updated to 08-17 version either.
>
> On Wed, Sep 26, 2012 at 11:05 AM, Tony Graziano
> wrote:
>>
>> I set the remote client to use UDP (polycom, bria and xlite). Anyone
>> e
I set the remote client to use UDP (polycom, bria and xlite). Anyone
else seeing this?
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Password or pin?
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1. I assume so.
2. Pdf is the default in 4.6.
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al" one that breaks the attended transfer. At this time that's the
> only call flow I've seen trouble with, but I haven't tried that many either.
>
> I'll include this info on the jira.
>
>
> - Jeff
>
>
>
> On Tue, Sep 25, 2012 at 12:44 PM, Tony Gr
es
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfou
elephone-event/8000
>>>> a=x-sipx-ntap:X192.168.54.46-[PUBLIC-IP];54
>>>
>>>
>>> Notice the c= and m= lines have changed. sipX is now telling the Adtran
>>> to send audio to 192.168.54.46, the IP address of the sipX instance itself.
>>
n Tue, Sep 25, 2012 at 12:37 PM, Jeff Pyle wrote:
> Tony,
>
> Yes, and yes.
>
>
> - Jeff
>
>
> On Tue, Sep 25, 2012 at 12:26 PM, Tony Graziano
> wrote:
>>
>> Is the adtran setup as an unmanaged gateway and on the same network as
&g
Is the adtran setup as an unmanaged gateway and on the same network as
sipx? If so that would be the correct configuration.
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The 3.3 and later firmware is not directly supported by sipx yet. Sear h
the lust archives for vvx500 to see how others have been able to configure
theirs.
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, I published another RPM minutes ago.
> Let me know if it gets updated
>
> Thanks
> George
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
> ___
tart.*
> ***
>
>
>
> ** **
>
> Hello,
>
> ** **
>
> for installing 4.6 please follow instructions from here:
>
> http://wiki.sipfoundry.org/display/sipXecs/Installing+4.6.0
>
> ** **
>
> Regards
>
> George
>
> ___
n use MRTG.
>
> On Mon, Sep 24, 2012 at 10:00 AM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> I think MRTG is a casualty. I'm not sure if it is being put back in or
>> not. With the "network queue" there are perhaps better and more co
y email
> and delete all copies.
>
> ___
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> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
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Only sipxbridge acts as a b2bua.
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I don't think I've heard any issues restoring 4.4 to 4.6, except yours.
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I suggested only restoring 4.2 to 4.6 was not something to assume would
work. Good luck.
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I'm am not sure restoring a 4.20 system to 4.6 would be advisable. I doubt
anyone is testing anything other than 4.4(latest) > 4.6(latest).
Perhaps you can create a 4.2 system and restore to that and update to 4.4
to grab a valid backup from.
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Explain the steps you have taken to get here.
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My point about superadmun ONLY had to do importing, not restoring.
Superadmin should be the one you had restored.
They logging in to the server at port 12000.
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I dont consider it strange exactly. It is more likely restarting presence
that needs to happen also.
Presence is still finicky like that.
Park really needs to move to a b2bua service too.
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VM expiration is also for unread voicemail in inbox. We don't want to
expire unread conferences.
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IMO this would be undesirable as users might need to reference conferences
for a long period.
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I'm see I g on 4.4 the call gets dropped and not into VM.
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Ask abou
t;> - Jeff
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ___
>>>
>>> sipx-users mailing list
>>>
>>> sipx-users@list.sipfoundry.org
>>>
>>>
gt; rebuilding sipxecs? Possibly an independent package, or is getting patched
> into sipXconfig the best way?
>
> Thanks,
>
> Shaun
>
> ___
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> sipx-dev@list.sipfoundry.org
> List Archive: http://l
updates
> without major functionality changes to SipX? Thank you.
>
>
>
>
>
>
>
> *Rick Heinlein*
>
>
>
> ___
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> sipx-users@list.sipfoundry.org
> List Archive: http://list.si
I've found that using DND on poly com phones results in calls not going to
voicemail unless this is left unchecked. I think VM should pickup but it
goes fast busy instead. If unchecked this rings silently at phone.
Anyone else think this is broken?
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telco.
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Using or developin
what kind of gateway/who is the telco?
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Please disregard this. The behavior is correct (4.4 sends PDF by default,
UNLESS the call is not completed, in which case the attachment is a TIFF
file, which is inconsequential since the testing was incorrectly reviewed).
On Tue, Sep 18, 2012 at 1:54 PM, Tony Graziano wrote:
> I installed
p://list.sipfoundry.org/archive/sipx-dev/
>
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ard.
A siptrace would be helpful.
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but must have reverted in a later patch. Is there a way to conform this? Do
we need a JIRA?
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Then how does your script discern a real sip call from a foreign system? It
must not be allowed since there is no phone registered.
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ecom Infrastructures *Royal Roads University*
>
> *T* 250.391.2600 ext. 4149
>
> 2005 Sooke Road, Victoria, BC Canada V9B 5Y2 *royalroads.ca*
>
> ****
>
> *LIFE.**CHANGING*
>
>
> On 2012-09-17, at 6:48 AM, "Tony Graziano"
Sounds like you are being bothered from the outside.
/var/log/sipxpbx
Is where logs are.
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Check the proxy and registrar logs. Also check CPU and ram/swap. The logs
may show a lot of call or registration attempts. If the phone are not
registering via the internet close off port 5060.
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(before I knew he was using openuc)
On Thu, Sep 13, 2012 at 5:26 PM, Douglas Hubler wrote:
>
>
> On Thu, Sep 13, 2012 at 4:07 PM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> so let us know what happends when you:
>>
>> /usr/bin/sipx-archi
http://track.sipfoundry.org/browse/XX-10437
On Thu, Sep 13, 2012 at 4:48 PM, Tony Graziano wrote:
> Unless there is no objection I will condense into a jira.
>
> On Thu, Sep 13, 2012 at 1:50 PM, Douglas Hubler wrote:
>
>> On Thu, Sep 13, 2012 at 11:38 AM, Tony Gr
Unless there is no objection I will condense into a jira.
On Thu, Sep 13, 2012 at 1:50 PM, Douglas Hubler wrote:
> On Thu, Sep 13, 2012 at 11:38 AM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> Using the below as a guide--What are the chances multiple package
s that you receive no errors in
> UI upon restore?
>
> There's CLI option too
>
> sipxecs
> sipxivr-archive --restore your-tar-file.tgz
>
> openuc
> openucivr-archive --restore your-tar-file.tgz
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>
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mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
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no dice.
>
>
> --
> *From: *"Tony Graziano"
> *To: *"Discussion list for users of sipXecs software" <
> sipx-users@list.sipfoundry.org>
> *Sent: *Thursday, September 13, 2012 2:07:59 PM
>
> *Subject: *Re: [sipx-users] 4.4.0 to 4.4.6
>
&
x27;t your own internal system you might
> > want to use 4.4.0 for
> > a production customer install.
>
>
> --
>
> ___
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> List Archive: http://list.sipfoundry.org/archive/sipx-use
Using the below as a guide--What are the chances multiple package updates
will send server profiles more than one time?
On Thu, Sep 13, 2012 at 11:31 AM, Douglas Hubler wrote:
> On Thu, Sep 13, 2012 at 11:13 AM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>&g
files!" as an automatic function? Does
this break anything else?
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Ask abo
les".
I am sure there are less obtrusive ways to do this depending on the
services being used, especially in a HA environment, but I think for those
of you who are test driving 4.6 right now this would be adequate for now.
Comments?
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> BTGSBS.btg-supplies.local, and should contain the following 0 attachment(s)
> . If any of these files are missing or if additional files are present we
> recommend you delete this message and contact us immediately.
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>
>
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somehow.
On Wed, Sep 12, 2012 at 9:13 AM, Douglas Hubler wrote:
>
>
> On Wed, Sep 12, 2012 at 8:05 AM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> I did that, in my case I moved the files, not copied them which if the DB
>> resides in mongodb direc
> configuration if service is disabled and enabled...
>
> Regards,
> Kumaran T
>
>
> On 9/12/2012 5:47 PM, Tony Graziano wrote:
>
> It would be helpful to understand this:
>
> When we disable a service like conference, does it remove the
> lines/services associated
it could be made a dependency from the sipx repo so
this does not bite a lot of folks?
On Wed, Sep 12, 2012 at 8:10 AM, George Niculae wrote:
> On Wed, Sep 12, 2012 at 3:05 PM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> I did that, in my case I moved the f
it could be made a dependency from the sipx repo so
this does not bite a lot of folks?
On Wed, Sep 12, 2012 at 8:10 AM, George Niculae wrote:
> On Wed, Sep 12, 2012 at 3:05 PM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> I did that, in my case I moved the f
to
re-create it. We should not clear data upon disabling the service. When I
try to create a user 222 it should throw an error that it exists.
On Wed, Sep 12, 2012 at 7:54 AM, Kumaran <
thiru.venkateshwa...@ttplservices.com> wrote:
> On 9/12/2012 5:20 PM, George Niculae wrote:
>
> O
mmitted the fix for wrong VM prompt after setting user caller-id
> in
> >> 4.6 or I can raise issue in Jira? Also please check the caller-id
> scenario
> >> in 4.4 build if you have(in openuc doesn't work but don't have sipxecs
> to
> >> verify it)...If
ailing list
> sipx-us...@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://ww
ailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://ww
t; sipx-dev mailing list
> sipx-dev@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Prof
indeed X-Assp-Envelope-From: is empty
is it a good/bad idea to show in ASSP log the From: value in case of the
envelope sender is not specified/empty ?
Graziano
> ASSP development mailing list schrei
> bt:
>> From: "Email Marketing"
>
>
> ASSP shows the envelop
gt;
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
> *Sent:* Tuesday, September 11, 2012 12:47 PM
> *To:* Discussion list for users of sipXecs software
> *Subject:* Re: [sipx-users] Calls dropping
> ** **
>
> Thanks again.
>
> ** **
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
> *Sent:* Tuesday, September 11, 2012 12:35 PM
> *To:* Discussion list for users of sipXecs software
> *S
e to the actual AA. When the DID was direct
> to the AA there was no “ringing” sound and they didn’t like it. Thus is
> life.
>
> ** **
>
> Anything else you need to know? I’m open to suggestions.
>
>
> Thank you!
>
> ** **
>
> *From:* sipx-users-boun...@list.sipfou
sible issue from my ITSP.
>
> Please point me in the direction or help me in any way possible. This is
> the only hangup I have had with this installation so far.
>
> Thank you in advance.
>
> ** **
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sip
_
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~~
header the from is clearly specified for example for the email
above it was the following
From: "Email Marketing"
Thank you
Graziano
--
Live Security Virtual Conference
Exclusive live event will cover al
etimes
make out what some people are calling about by glancing at the text, others
not so much.
I still chuckle a lot because google translates "Graziano" as "crazy, I
know".
On Tue, Sep 11, 2012 at 8:20 AM, Nathaniel Watkins <
nwatk...@garrettcounty.org> wrote:
> Agr
There are examples of this for FreeSWITCH in their forum and wiki.
--
~~
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-graziano/14/4a6/7a4
Ask about
ress 0xfff5
> ) at HEPDao.cpp:61
>
>
> The crash is happening when sipxhomer tried to connect to mysql.
>
>
> Tony,
>
> Thansk for verifying. Will test with 32 bit as soon as time permits.
>
> J
>
>
>
>
>
>
>
>
>
>
>
> On 09/06/2012 11:10
n to_string
return self.val['_M_dataplus']['_M_p'].lazy_string (length = len)
RuntimeError: Cannot access memory at address 0xfff5
) at HEPDao.cpp:61
#3 0x0806d0ea in main (argc=7, argv=0xb724) at main.cpp:40
"That's all she wrote!"
On Thu, Sep 6, 2012 at 11:13 AM
I did. That was the entire output.
On Thu, Sep 6, 2012 at 11:10 AM, George Niculae wrote:
> On Thu, Sep 6, 2012 at 6:07 PM, Tony Graziano
> wrote:
>> it is, must have been in the wrong directory after all. sipxhomer is there.
>>
>> [Thread debugging using libthread_db
iculae wrote:
> On Thu, Sep 6, 2012 at 5:58 PM, George Niculae wrote:
>> On Thu, Sep 6, 2012 at 5:50 PM, Tony Graziano
>> wrote:
>>> this is my input in gdb:
>>>
>>> run --log-file /var/log/sipxpbx/sipxhomer.log --config-file
>>> /etc/sipxpbx/sip
.
Use the "file" or "exec-file" command.
***
On Thu, Sep 6, 2012 at 10:48 AM, George Niculae wrote:
> On Thu, Sep 6, 2012 at 5:46 PM, Tony Graziano
> wrote:
>> -This is the 3rd install on 32 bit, they all broke. It broke last
>> month after an update.
>&g
machine and got the same behavior? Most developers are in 64
> bit environment so I wont be surprised if there are some surprises with 32
> bit boxes. A back trace as advised by George might bring us closer to the
> truth.
>
> Joegen
>
>
> On 09/06/2012 08:48 PM, Tony Grazia
at 3:51 PM, George Niculae wrote:
>> On Thu, Sep 6, 2012 at 3:48 PM, Tony Graziano
>> wrote:
>>> We still see this on 32 bit ISO installs with the latest updates:
>>>
>>> kernel: sipxhomer[633]: segfault at 10009 ip 07035bc6 sp bfdea274
>>> erro
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