On Thu, 6 May 2010 18:10:46 -0400, Tony Graziano wrote:
> Check the sipxconfig.log
In that log, there are a lot of references to left over certificate files
around the system which still have the old name.
"2010-05-06T20:22:12.437000Z":418:JAVA:INFO:uc.newdomain.net:pool-2-thread-1::XmlR
On Thu, 6 May 2010 18:10:46 -0400, Tony Graziano wrote:
> Check the sipxconfig.log
In that log, there are a lot of references to left over certificate files
around the system which still have the old name.
"2010-05-06T20:22:12.437000Z":418:JAVA:INFO:uc.newdomain.net:pool-2-thread-1::Xml
On Thu, 06 May 2010 16:42:13 -0500, Josh Patten wrote:
> You can run sipxecs-setup to try to rewrite all of those settings that
> somehow got messed up. As far as I know it doesn't overwrite the
> database (someone correct me if it does).
Yup, I've done that as well. This time, went with the reboo
I'm at a loss so thought I would post for some help.
I changed the domain name of a sipx 4.2 server.
I ran sipxecs-setup and did it there as well.
I rebooted the server and used the following commands to check if everything is
running.
[r...@uc ~]# /etc/init.d/sipxecs status
Checking sipxsupervi
> Just change the redirect link in the Apache configuration
> (/usr/share/www/doc/index.html) to port 443 instead of 8443, and
> then use iptables to "forward" all incoming TCP traffic on port 443 to
> 8443...
> Works for me but I'm not sure it's the best way. Anyone else care to
> comment what I
rs-
>> boun...@list.sipfoundry.org] On Behalf Of Staffan Kerker
>> Sent: Thursday, May 06, 2010 12:01 PM
>> To: m...@grounded.net
>> Cc: sipx-users
>> Subject: Re: [sipx-users] Using port 443 instead of 8443
>>
>> Hi
>>
>> I actually did a work arou
Can sipx be run on port 443 instead of 8443?
And if so, what would require changing on the system to make it so.
Thanks.
Mike
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Unsubs
On Wed, 5 May 2010 10:39:03 -0400, Picher, Michael wrote:
> I mean just won't work...
K, just checking, I especially don't want to kill a 650 :).
Thanks for the input.
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at, except it used AA
> batteries.
>
> On 05/04/2010 09:35 PM, m...@grounded.net wrote: Yup, no audio and can't
> seem to find a power supply for the amp. On Mon, 03 May 2010 22:22:08 -
> 0500, Josh Patten wrote: > At the worst you'd get no sound. At the best,
> i
On Tue, 4 May 2010 23:35:34 -0400, Tony Graziano wrote:
> make sure you hit the menu options in the polycom to tell it what type of
> headset it is.
Oh, I never did do that. I'll take a look at that also, thanks.
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On Tue, 04 May 2010 21:52:55 -0500, Josh Patten wrote:
> If you can get your hands on one it should work. Just plug the one that
> goes to the phone in the headset jack of the amp then just press the
> headset button on the phone when you want to use it (assuming you find the
> power supply for it)
try. Just wanted to make sure nothing
would blow up.
Mike
> http://www.jabra.com/NA-US/headsetsolutions/Pages/JabraAudioEnhancers.aspx
> (GN-1200)
>
> On 05/03/2010 10:18 PM, m...@grounded.net wrote:
>> On Mon, 03 May 2010 22:06:39 -0500, Josh Patten wrote:
>>
&
http://www.jabra.com/NA-US/headsetsolutions/Pages/JabraAudioEnhancers.aspx
> (GN-1200)
>
> On 05/03/2010 10:18 PM, m...@grounded.net wrote:
>> On Mon, 03 May 2010 22:06:39 -0500, Josh Patten wrote:
>>
>>> Is there no way you can try them out? Looks pretty standard, in fact
On Mon, 03 May 2010 22:06:39 -0500, Josh Patten wrote:
> Is there no way you can try them out? Looks pretty standard, in fact you
> could try ditching the little amp altogether and plug straight in the
> headset port. Just tinker with one if you can.
Sure can, just seem to recall someone saying th
he connectors.
>
> On 05/03/2010 06:25 PM, m...@grounded.net wrote:
>> I have a bunch of Lucent KS23822-L26LU or TR3566 headset adapters for an
>> Avaya IP 400 phone system. Before I junk them, I was wondering if anyone
>> might known if these could work with the polycom 3
I have a bunch of Lucent KS23822-L26LU or TR3566 headset adapters for an Avaya
IP 400 phone system. Before I junk them, I was wondering if anyone might known
if these could work with the polycom 330 or 650 phones.
Worth a shot :)
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sipx-users maili
> Not sure what changes you are making to the web pages, but there are methods
> of making changes via CSS and others that allow those changes to remain and
> appear on reboot. It's documented in the wiki. DO a search on sipXecs web
> template, that should locate the relevant pages.
We probably
rence wrote:
>�On Sun, 2010-05-02 at 19:39 -0500, m...@grounded.net wrote:
>�
>>�Is there any way in 4.2 of preventing the web structure overwrite upon
>>�reboot/restart?
>>�
>�It's not clear what you're referring to here... can you be more s
>Is there any way in 4.2 of preventing the web structure overwrite upon
>reboot/restart?
Does part of the web structure run in some sort of cache?
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That was indeed the problem, didn't have a wildcard so quickly bought one for
the sipx server and it went in fine.
Thanks again.
On Sun, 2 May 2010 19:41:57 -0500, m...@grounded.net wrote:
> Could someone post how they went about installing godaddy ssl certs on
> their 4.2 server.
I guess I don't have a wildcard cert. Off to buy another one for now.
On Sun, 2 May 2010 20:44:14 -0500, m...@grounded.net wrote:
> Ok, thanks again, that worked just fine this time around.
>
> Mike
>
>
> On Sun, 02 May 2010 20:04:23 -0500, Josh Patten wrote:
>>
Maybe something is still left over. Checking by connecting gives me,
sx.mydomain.com:8443 uses an invalid security certificate.
The certificate is only valid for the following names:
www.mydomain.com , mydomain.com
(Error code: ssl_error_bad_cert_domain)
Yet, should work as the domain is co
ce.
>
> On 05/02/2010 08:03 PM, m...@grounded.net wrote:
>>> If you have the key and cert then you don't need to worry about the CSR.
>>> That's the beauty of having both.
>>>
>>> The CSR is the certificate signing request and that generates a
> If you have the key and cert then you don't need to worry about the CSR.
> That's the beauty of having both.
>
> The CSR is the certificate signing request and that generates a private key
> for you. If you already have the private key then you don't need to do a
> CSR.
>
> http://en.wikipedia.
atten wrote:
> With a wildcard cert you have to have the .key that came with the
>
> certificate. Other than that it should work fine.
>
> On 05/02/2010 06:52 PM, m...@grounded.net wrote:
>> I have a cert which is www.domain.com and *.domain.com which is being
>> used on
Is there any way in 4.2 of preventing the web structure overwrite upon
reboot/restart?
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gt;> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
>> boun...@list.sipfoundry.org] On Behalf Of m...@grounded.net
>> Sent: Sunday, May 02, 2010 7:53 PM
>> To: sipx-users
>> Subject: [sipx-users] ssl cert
>>
>> I have a cert which is www.domain.
I have a cert which is www.domain.com and *.domain.com which is being used on a
web server which has the same domain name as the sipx server does. Could I
install this cert on sipx safely or should I purchase another cert? And, in
4.2, are ssl certs easier to install?
Thanks.
Mike
__
Question: If many of you already know or figured out how to get alternative
notifications working, that must mean there is either more notes somewhere or
that some of you are part of the dev team, or that you're just plain genius.
I'm not having any luck getting this to work and short of begging
>
> forward :-) !!
>
>> -Original Message-
>> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
>> boun...@list.sipfoundry.org] On Behalf Of m...@grounded.net
>> Sent: Friday, April 23, 2010 12:39 PM
>> To: sipx-users
>> Subject: Re: [sipx-users]
Is there a chance that someone is going to write a little howto for this? :).
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Anyone else get this spam from a sipx addy?
Alan\
On Thu, 22 Apr 2010 18:52:05 -0600, Alan\ wrote:
> http://basecomputer.org/home/index.php
>
> ___
> sipx-users mailing list sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/arc
>> As there is no sipXconfig support for configuring this yet, the default
>> format for the main email address is Full, and for the alternate email
>> address is Brief at the moment.
I thought perhaps this actually meant that alternate email was the alternate
notification but it's not.
At a loss
> Place the file in /etc/sipxpbx/sipxivr Josh Patten Assistant Network
Yup, I have, and permissions/ownership are fine too.
>sipXivr will look for {prefix}/etc/sipxpbx/sipxivr/EmailFormats.properties
>(including all EmailFormats_{locale}.properties variants) first, and if
>not found will use the
I never restarted any services after putting the file in there, is that a
requirement for sipx to find it?
On Thu, 22 Apr 2010 13:40:42 -0500, m...@grounded.net wrote:
> Do you mean in the Users section of the GUI, in Unified Messaging, the
> Full, Medium or Brief options?
>
> I
st.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> m...@grounded.net
> Sent: Thursday, April 22, 2010 8:36 AM
> To: sipx-users
> Subject: Re: [sipx-users] Custom Notifications
>
> I am still missing something and this new feature doesn't have
How the heck do you guys figure these things out without documentation unless
you are part of the dev team as well?
I've looked at every file in /etc/sipxpbx and see plenty of email and
notification options but none which seem to affect which notice I can send out,
meaning using the file I plac
default settings. After you have
> created it THEN you can modify it to your needs. Once you get it in
> place it will make sense
>
> m...@grounded.net wrote:
>> Thanks also Tony.
>>
>> Ok, so I guess I had found this yesterday but it was jiberish. Still is
>> t
Sorry, I forgot to include that you would still have to log into your system to
see the log. Or perhaps send all of your syslog output to another server.
On Sun, 18 Apr 2010 09:51:28 -0700, Charles wrote:
> As a linux novice running sipxecs on a machine that does not have a monitor
> attached I
You could output the console to syslog. Take a look at your syslog config in
/etc/
On Sun, 18 Apr 2010 09:51:28 -0700, Charles wrote:
> As a linux novice running sipxecs on a machine that does not have a monitor
> attached I had a question regarding monitoring the server.
>
> I frequently login
you have
> created it THEN you can modify it to your needs. Once you get it in
> place it will make sense
>
> m...@grounded.net wrote:
>> Thanks also Tony.
>>
>> Ok, so I guess I had found this yesterday but it was jiberish. Still is
>> today.
>> I though
Graziano wrote:
> On Sat, Apr 17, 2010 at 5:45 PM, m...@grounded.net
> wrote:
>
>> Thanks Josh.
>>
>> I'm getting a 404 on this link though.
>>
>>> The requested resource cannot be found.
>>> Path
>>> "main/sipXivr/pocosan
by default. Follow the instructions I posted earlier
> to create the file and fill it with the default settings. After you have
> created it THEN you can modify it to your needs. Once you get it in
> place it will make sense
>
> m...@grounded.net wrote:
>> Thanks also Tony
Thanks also Tony.
Ok, so I guess I had found this yesterday but it was jiberish. Still is today.
I thought it would be a simple to understand file that has the current method
and sections for entering my own instead.
I suspect there will be others asking the same things :).
I see these items i
lso, make sure the permissions are correct
> (should be owned by sipxchange:sipxchange I think)
>
> m...@grounded.net wrote: > Still no luck finding out how to do this. On
> Fri, 16 Apr 2010 14:22:40 -0500, m...@grounded.net wrote: > > sipXivr
> will look for {prefix
Still no luck finding out how to do this.
On Fri, 16 Apr 2010 14:22:40 -0500, m...@grounded.net wrote:
>> sipXivr will look for {prefix}/etc/sipxpbx/sipxivr/EmailFormats.properties
>>
>> (including all EmailFormats_{locale}.properties variants) first, and if
>> not fo
>sipXivr will look for {prefix}/etc/sipxpbx/sipxivr/EmailFormats.properties
>(including all EmailFormats_{locale}.properties variants) first, and if
>not found will use the default formats defined in
>org.sipfoundry.voicemail.EmailFormats.properties (in the sipXivr.jar file).
>Per user selectable
> Yes, go to your etc/yum.repos.d/sipXecs.repo file to make edits.
> Change all lines that have 5.2 in it to 5
Didn't work for me when I tried that yesterday so I just enabled the one in the
repo file and was able to update. Any problem with that? Two of the servers
I've done are working fine i
Earlier, I posted that we started noticing clicks and pops that we thought
hadn't been there before.
Strangely, I upgraded one of the servers that I reported earlier and after the
upgrade, it's clear as a bell now.
I'll upgrade another and see what happens.
_
Should I start a new thread since this isn't memory related.
On Thu, 15 Apr 2010 11:06:43 -0500, m...@grounded.net wrote:
> I seem to have an intermittent problem with a blade in a BladeCenter.
>
> DIAGS: 001-250-001-20100414-56-Core System : Failed ECC_MEM&1 (CPU 1:
> Sin
Forgot to mention... this is using sipx only. No gateways, nothing else, just
sipx and a phone on the LAN.
On Thu, 15 Apr 2010 12:10:36 -0500, m...@grounded.net wrote:
> On Thu, 15 Apr 2010 11:16:51 -0500, Matthew Kitchin (public/usenet) wrote:
>
>> Here is someone else who was ex
On Thu, 15 Apr 2010 11:16:51 -0500, Matthew Kitchin (public/usenet) wrote:
> Here is someone else who was experiencing a clicking. Different
> scenario, but maybe you can find a common cause...
Saw that thread, haven't found an answer.
The testing we've done so far is to turn ACD off on two blade
I seem to have an intermittent problem with a blade in a BladeCenter.
DIAGS: 001-250-001-20100414-56-Core System : Failed ECC_MEM&1 (CPU 1: Single
Bit Error Not Set
I'll change the memory out but am wondering what else I should look at. When
recording messages, we are hearing a sort of clickin
On Wed, 14 Apr 2010 16:09:16 -0400, Tony Graziano wrote:
> I don't prefer beer in my coffee. Something stronger and darker from a
> barrel out of kentucky.
This morning, just the thought of that makes my head spin.
We spent last evening on part two of a beer making class at a local distillery.
P
at that.
>>
>> -Original Message-
>> From: sipx-users-boun...@list.sipfoundry.org
>> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
>> m...@grounded.net
>> Sent: Wednesday, April 14, 2010 10:24 AM
>> To: sipx-users
>> Subject: Re:
rt's sense of humor..and
> clever at that.
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> m...@grounded.net
> Sent: Wednesday, April 14, 2010 10:24 AM
> T
I
> would say more like '+30' :)
>
>>
>> On 4/14/2010 11:11 AM, m...@grounded.net wrote:
>>> Is there some way of making the lists display more items in
>> the Users listings for example?
Is there some way of making the lists display more items in the Users listings
for example?
Only 20 at a time show up, I would prefer more.
Thanks.
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One of the very first things would be to simply change the color scheme so that
light text is not on light background.
Who's idea was that? :)
On Tue, 13 Apr 2010 20:01:27 -0400 (EDT), Administrator wrote:
>
>
> Content-Type: text/plain;
> charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Org
> Isn't the 330 a non-HD unit? I have 650's and 335's and calling between a
> 335 and a 650 results in the little 'HD' logo on the line appearance and
> great sounding audio (handset or speakerphone).
Yes, it's non HD. My test was just to see if the two polycom's talking together
would sound any
>- If you decide you want to manually configure because sipxecs doesn't
> cover the way you want to use it you may want to kill yourself with a dull
> butter knife (the configuration is not obvious, and the documentation is of
What I found in learning about these things was that the initial le
I have a number of 330's and a while back, I bought a 650 so that I could test
HD voice.
I've not bought another 650 so guessing I'm not getting the full HD experience
but figured I should get some darn good quality even 650 to 330. Both of the
phone sound incredible on hands free but pretty muc
I asked for mediant firmware many times and I believe that people are very
hesitant to help with firmware.
The most common reply was: "Search the Internet and you will find"... And I did
in fact :).
On Tue, 06 Apr 2010 08:29:19 -0700, Johan Reinalda wrote:
> That is the conclusion we have come
On Mon, 5 Apr 2010 14:11:32 -0400, Tony Graziano wrote:
> Not if they've forgotten it.
Darn, that's true. Feature request!
> Cept they can easily change it from the GUI though.
>> Sounds like a feature request to me:
>> Forgot you pi? Click here to have a tempirary pin emailed to the address
>>
Cept they can easily change it from the GUI though.
On Mon, 5 Apr 2010 13:49:40 -0400, Tony Graziano wrote:
> Sounds like a feature request to me:
>
> Forgot you pi? Click here to have a tempirary pin emailed to the address on
> record.
>
> Tony Graziano, Manager
> T
On Mon, 05 Apr 2010 10:43:43 -0700, Todd Hodgen wrote:
> If you are just making changes to the account, you don't have to change the
> Pin number. Are you possibly changing their line number?
Oops, my mistake. Guess I started changing the PIN automatically, thinking it
would prompt me anyhow.
Th
Don't suppose there is a trick to NOT having to change the PIN number when one
changes a user account?
Rather frustrating to users if we need to change something then have to tell
them their PIN was changed on them.
Mike
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On Tue, 30 Mar 2010 18:48:42 -0700, Todd Hodgen wrote:
> Are you talking about on your local network? If you are, that gets defined
> in sipxecs configuration so it works with what you define.
Yes, just wondered if sipx might have any issues with nets larger than /24.
___
> I have not had any issues using a class B.
Nor I, just wondered if there might be some anomaly I wasn't aware of. Thanks
for the input everyone.
Mike
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So, sipx doesn't really have any issues with mask size then. That's what I was
wondering about.
On Tue, 30 Mar 2010 13:27:58 -0400, JOLY, ROBERT (ROBERT) wrote:
>> Catchy subject eh :).
>>
>> Anyhow, in the Internet Calling section, then then Intranet
>> Subnets section, in my case, I have 192.
I know that devices must be in the proper network to be able to talk to each
other etc.
What I am wondering is if sipx has any problems with a larger mask than the
usual 24 that most folks are on.
Mike
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On Tue, 30 Mar 2010 13:20:38 -0400, Tony Graziano wrote:
> The mask for your subnet needs to match.
> What kind of issue are you having?
No issues, just wanting to better understand this part.
Are you saying that everything to do with sipx should be in the same subnet, as
in /24 or that a /16 is
Catchy subject eh :).
Anyhow, in the Internet Calling section, then then Intranet Subnets section, in
my case, I have 192.168.0.0/16.
Is there any chance that sipx might be unhappy with this mask and might prefer
a 24, even keeping SIP devices in the same segment rather than a 16?
Mike
_
> The sensitivity to packet loss or other network effects (packet
> reordering or jitter, for example) are highly dependent on the
> capabilities of the endpoints. There are many different strategies for
> compensating for these things, some of which are quite sophisticated and
> leverage the way
> Make sure you turn on debug logging on the sipXproxy and the sipXregistrar
> before reproducing the failing scenarios and taking the snapshot. Also,
> clearly take note of each steps involved in each scenario, i.e. who calls
> who and what the results are at each end.
Ok, so first, the remote u
Got the tests done. Where do I send this so that I'm not posting all sorts of
information I prefer not having on the net? :)
On Thu, 25 Mar 2010 17:51:52 -0400, JOLY, ROBERT (ROBERT) wrote:
>> The remote can call out. We tested by making a call to a cell phone.
>>
>> Using an internal to call t
Sounds like a plan, will get on that later on then.
On Thu, 25 Mar 2010 17:51:52 -0400, JOLY, ROBERT (ROBERT) wrote:
>> The remote can call out. We tested by making a call to a cell phone.
>>
>> Using an internal to call the remote leads to ringing,
>> nothing else, not even vm.
>> Using an exter
The remote can call out. We tested by making a call to a cell phone.
Using an internal to call the remote leads to ringing, nothing else, not even
vm.
Using an external phone, cell phone, calling the remote, leads to a fast busy.
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> This is a 'normal error'. The sipXproxy will periodically (every 20 secs.)
That's what we gathered but wanted to confirm, thanks.
> Stay away from Internet calling if you can avoid it. Please visit
Ok, will do that.
> Check out "Problem #2" in link above and check that the IP addresses tha
When I take a phone and plug it into a wireless remote network connection, the
following keeps showing up in sipXproxy.log.
"2010-03-25T19:48:50.052314Z":3099:INCOMING:INFO:uc.domain.com:SipClientUdp-8:B78FCB90:SipXProxy:"Read
SIP message:\nRemote Host:xxx.xxx.xxx.215 Port: 5060\nSIP
> Any packet loss will result in a garble. A little bit more makes someone
> into darth vader. A little bit more means your call just got blasted into
> smithereens by the death star.
Hehe, Darth Vader! ROFL.
Thanks.
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>
>
> I can only draw stick people animations, and they are not pretty, so I'll
> leave it at that.
>
>
>> Todd Hodgen wrote: > My vote is its UA to UA since the Bridge is not
>> involved. -Original Message- From: sipx-users-
>> boun...@list.si
ROFL, the beloved forums that some of us wanted.
First of all, can't read them very well. Who chose a light colored text against
a light colored background? Ow!
But what's extra funny is that now there is double the noise on search engines
when trying to find sipx related information.
The sam
On Mon, 22 Mar 2010 19:21:28 -0400, Picher, Michael wrote:
> Two remote users (each behind NAT) through sipX is an interesting question
> as to whether the call is phone to phone or phone to pbx to phone. I
> believe the latter may be true. Ranga can answer that.
In my application, it would be p
On Mon, 22 Mar 2010 18:43:29 -0400, Picher, Michael wrote:
> True internally on a network, but if you are talking about a sip trunk the
> call will hairpin on the PBX and require 2x the bandwidth.
I should have been more specific but yes, I mean two remote users, not two
local users, over the Int
ipfoundry.org [mailto:sipx-users-
>> boun...@list.sipfoundry.org] On Behalf Of m...@grounded.net
>> Sent: Monday, March 22, 2010 5:05 PM
>> To: sipx-users
>> Subject: [sipx-users] Packet loss Question
>>
>> I've noticed that one of my providers keeps discon
I've noticed that one of my providers keeps disconnecting for a few seconds at
a time.
I was wondering what effect this has on sipx users.
For example, when using SIP, I can see how important the connection needs to
be. But once end points are talking together and media is flowing between them,
> You mean terminating a user agent's Registration ahead of its scheduled
> expiry? If so, the only non-hacker way to achieve this is for that user
> agent to terminate its own registration by sending a REGISTER with expiry=0.
Yes, that's what I mean. So I guess it can be done, but the user needs
Is there any way at all of quickly expiring a user rather than having to wait?
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Cool, thanks :)
On Thu, 18 Mar 2010 19:34:42 -0400, Tony Graziano wrote:
> Correct amundo.
>
>
> Tony Graziano, Manager
> Telephone: 434.984.8430
> Fax: 434.984.8431
>
> Email: tgrazi...@myitdepartment.net
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> T
I had not done an update in some time and am not seeing any.
Can anyone confirm that I am at the latest stable?
No software updates available
The package information was last updated Mar 18, 2010 6:28 PM.
The system is currently running sipxcommons 4.0.4-017289
_
; in no GUI.
>
> I do all my asterisk work without GUI (text files only).
> Josh Patten Assistant Network Administrator Brazos County IT Dept. (979)
> 361-4676
> On 3/16/2010 10:06 AM, m...@grounded.net wrote: > The simplest way is to
> start with a pbx. For example, set up
k? I get
>
> IAXmodem to work but hylafax just doesn't seem to want to do anything.
>
> Josh Patten
> Assistant Network Administrator
> Brazos County IT Dept.
> (979) 361-4676
>
>
> On 3/16/2010 9:35 AM, m...@grounded.net wrote:
>> Yes, I've go
one sipX account to let FAXVOIP register exp : PSTN -
> -m1k-->sipxecs-->FAXVOIP server )
>
>
> m...@grounded.net wrote: > http://sipx-
> wiki.calivia.com/index.php/HowTo_configure_AudioCodes_SIP_Gateway_with_sipX
> I have been working on a mediant 2k (5.6) configuration
30 -0500, Scott Lawrence wrote:
>
>> On Mon, 2010-02-15 at 21:26 -0600, m...@grounded.net wrote:
>>
>>> In asterisk, one of the main suggestions for security is to make your
>>> SIP user names different than your extensions.
>>>
>> A pointless a
On Mon, 15 Feb 2010 19:30:21 -0800, Todd Hodgen wrote:
> Mike, You can use a name for your line, and then put the extension number in
> as an alias.
Oh darn, yes, that's true. Wasn't thinking :).
Thanks for the input.
___
sipx-users mailing list sipx-u
In asterisk, one of the main suggestions for security is to make your SIP user
names different than your extensions.
In sipx, this does not seem to be an option. Anything to be concerned about in
terms of hacking since all of our users are remote.
Mike
__
The point is that it is known to be a good application but it has not, so far,
worked out for me. That doesn't imply that it's not valid.
On Fri, 05 Feb 2010 21:31:16 -0500, Michael Scheidell wrote:
> On 2/5/10 1:50 PM, m...@grounded.net wrote: > OpenSBC is open source and
>
OpenSBC is open source and allows you to load balance using dns srv.
Only problem is that things can get troublesome. I fought gremlins for some 6
months and have yet to figure out why I was having so many remote problems.
Constantly problems, but they might have been related to the fact that the
..@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> m...@grounded.net
> Sent: Saturday, January 23, 2010 3:51 PM
> To: sipx-users
> Subject: [sipx-users] RTP Ports
>
> I've taken my SBCs down for testing and all of my remotes have
> 1
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