I tested using a real email address and a new account but with one of the
voip numbers, same problem.
I'll try a new DID provider next.
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-user
>Not at all sipx related. Get thee to a carrier who has better peering
>for toll calls or suffer silently.
BTW, one interesting reason I read about was that supposedly CL doesn't like
'free email' accounts.
Guess I'll have to test that by posting something and using one of the voip
DID's suppose
h
as voip,ms and flowroute which doesn't have this problem.
Mike
On Tue, Jan 8, 2013 at 12:50 PM, wrote:
> In fact, it’s likely not sipx related at all but I figured I’d ask here.
>
> We get complaints now and then that certain calls never work. One good
> example would b
In fact, it’s likely not sipx related at all but I figured I’d ask here.
We get complaints now and then that certain calls never work. One good example
would be when people post stuff on Craigslist and that CL complains that
certain voip numbers don’t work. In fact, it is looking like most don’t
>Yep, as Laurentiu mentioned you need to enable registrar and proxy on
>that machine. Go to Servers > Telephony tab and choose services you
>want to run on the box (registrar, proxy, voicemail)
I like that services are modularized. It's a nice way of managing servers,
turning things on and off to
>You should enable 'SIP Registrar' service in order to have this page
Got it, thanks.
I’m not sure it’s going to be a good idea to go to 4.6.0 on a remote system
because of it’s many changes. Might be safer to rebuild a 4.4.0 then get on
4.6.0 again in a few
>Password or pin?
No bug, just mixed up my passwords due to the new Voicemail PIN option.
Thanks for the input on the other two items.
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In
I restored a newly installed 4.6.0 using a 4.4.0 backup.
The only options I picked when restoring was to keep the same host/domain name,
nothing else.
In testing, I was trying to log in as a user but could not. I looked at that
users account and it’s password looked to be the same as on the 4.4
Couple of quick questions...
1: Is the ' EmailFormats.properties' file still needed in the
/etc/sipxpbx/sipxivr directory for custom email text?
2: How does one set incoming faxes to PDF?
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List
a few days of space.
I've been searching the wiki and can't find anything about the Registrations
screen being removed or getting at it if it's still there.
Other than that, everything else looks familiar enough at the moment.
Mike
_
>Well we're running 4.6 on eZuce for 2 months now and it works just
>fine. Of course there are updates / bug fixes but nothing to affect
>core functionality. Hope you'll get back to 4.6.0 really soon
I'm in the middle of a move which is why I thought building a 4.4.0 system
would be the safest.
I
>Well we're running 4.6 on eZuce for 2 months now and it works just
>fine. Of course there are updates / bug fixes but nothing to affect
>core functionality. Hope you'll get back to 4.6.0 really soon
I could use 4.6.0, just a little nervous that since the server will be
remote to us, it could mea
little time.
Mike
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
>Or use sipxecs-setup --reset-all that should automatically drop all dbs
The restore seemed to work the second time around so was able to spend a little
time with 4.6.0. However, I need to convert this server to a 4.4.0.
This was a fresh Centos 6.3 install with sipx installed via repo. Other than
Sent too quickly... after trying the command you sent me, then I went to gui to
log in and logged in with no password.
All users and settings minus domain name from the previous server appear to be
restored onto new 4.6.0 server.
Everything appears to be as it should be.
From: George Niculae
S
the command you gave me; /etc/init.d/sipxconfig reset-admin
log4j:WARN No such property [conversionPattern] in
org.sipfoundry.commons.log4j.SipFoundryLayout.
[root@sx0 ~]#
Waiting for input.
Mike
From: George Niculae
Sent: Sunday, September 23, 2012 4:11 PM
To: m...@grounded.net ; Discussion
I rebuilt it so it’s a new server but sure, I’ll give it a try and post what
happens.
From: George Niculae
Sent: Sunday, September 23, 2012 4:11 PM
To: m...@grounded.net ; Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6.0 superadmin password lost
Maybe you hit a bu
; Restore is supported, to reset password do a
>>
>> /etc/init.d/sipxconfig reset-admin
>>
>> Use help to get the full list of options
>>
>> George
>>
>>> Of course, that gets me wondering out of curiosity...
>>>
>>&g
ry. If that’s not something which is
>>> working yet, no big deal.
>>>
>>
>> Restore is supported, to reset password do a
>>
>> /etc/init.d/sipxconfig reset-admin
>>
>> Use help to get the full list
Thank you for the good tips. I’ll keep these for future reference and give
4.6.0 another try soon.
Mike
From: George Niculae
Sent: Sunday, September 23, 2012 1:13 PM
To: m...@grounded.net ; Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6.0 superadmin password lost
sity...
>
> Can I now yum remove the sipx installation, then update the repo to 4.4.0 and
> re-install? I would delete the postgres database and /etc/sipxpbx.
>
>
>
> Mike
>
>
>
>
>
> ~~
> Tony Graziano, Manager
> Telephone: 434.984.84
>Restore is supported, to reset password do a
>/etc/init.d/sipxconfig reset-admin
>Use help to get the full list of options
>George
Wonderful, thanks very much. I’ll keep playing with it.
Mike
> Of course, that gets me wondering out of curiosity...
>
> Can I no
s database and /etc/sipxpbx.
Mike
~~
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 our Interne
4.6.0 along with commands that don’t seem to work on 4.6.0. I have not yet
found information on how to restore the superadmin password on a 4.6.0 system.
Mike
From: Tony Graziano
Sent: Sunday, September 23, 2012 10:36 AM
To: m...@grounded.net ; Sipx-users list
Subject: Re: [sipx-users] 4.6.0
te the superadmin account password on 4.6.0. All of the posts
I’ve come across relate to older systems so none of the commands are the same.
Does anyone know how to set a new password for superadmin other than hacking
the database table itself?
password for superadmin other than hacking
the database table itself?
Mike
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Is the wiki down?
Can’t seem to get a response.
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
>Yes, if this isn't your own internal system you might want to use 4.4.0 for a
>production customer install.
Sounds like 4.6.0 is solid enough to install an instance and start putting it
through some testing.
I’ll try to get a system up and see how things go.
Thanks Todd and Mike.
You’re right, that was mentioned before but my notes didn’t make any sense .
I’ll update that now and will install a 4.4.0 for now.
Also, it’s great to know that 4.6.0 will allow a restore of the old system.
That’s such a great advancement for sipx.
Mike
From: Michael
Will 4.4.0 upgrade to 4.6.0 when 4.6.0 is stable ready?
In other words, I need to build a system so would probably want to use 4.4.0
right.
Mike
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org
That’s what I figure but wondered if some had dealt with this before since it’s
coming up now and then.
Thanks.
Mike
From: Tony Graziano
Sent: Tuesday, July 24, 2012 10:10 AM
To: m...@grounded.net ; Sipx-users list
Subject: Re: [sipx-users] V.150 MoIP Modem
You are better off just getting
come across this and how have you dealt with the
problem without resorting to standard POTS lines.
Thanks.
Mike
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Does anyone know what the expected release date is for 4.6?
Mike Burden
[cid:image002.gif@01CD4234.AA319080]
Lynk Systems, Inc
e-mail: m...@lynk.com<mailto:m...@lynk.com>
Phone: 616-532-4985
<><>___
sipx-users maili
ts. With that experience, they might
also be a good choice for cloud-based servers. I haven't run a box
at Rackspace in years, though.
--
Mike Pinkerton
On 23 May 2012, at 20:44, Joegen Baclor wrote:
> You can get a bare metal box at this price level from iWeb. I've
>
Tony:
That volume is way beyond what we will do. Our inbound faxes are
very important, 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
or she would recommend? If so, are there any peculiar config
settings required to work with that provider?
Thanks.
--
Mike Pinkerton
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
That is a very kind way of saying "foolish move".
Before stepping all the way back to Fedora 14, I might try another
foolish move -- installing 4.4 stable on Fedora 16 with a couple of
Fedora 14 packages thrown in to cover the missing dependencies. If
that doesn't work, then I'll go back t
t location today and don't have the exact
log message.
My questions:
1. What is the best way of alerting the developers to this
regression in the latest snapshot?
2. What is the best way to get a stable set of packages onto a stock
Fedora 16 box?
Thanks.
--
Mike
PS: My first
Unfortunately, we were never able to get the Edgewater device AirBand
installs to work properly with
We ended up switching away from SipXecs and things are working great now.
We are not 100% sure where the problem was, nor did we have much time to
resolve the issue. AirBand was very helpful durin
27;re testing in
production tomorrow. I needed both domains listed as
Intranet Domains under System > Internet Calling. After
adjusting some other settings, call control was working, but
there was no audio. I'm guessing sipX was trying to do some
sort of translation and sending the
I'm double checking DNS now. Interesting, we have an FXS
ATA attached to domain A that domain B has no trouble
transferring. The transfers only fail on a Polycom phone.
Mike
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http:/
e has run into a similar situation or
found a solution.
mailto:2...@domaina.org calls mailto:5...@domainb.org.
mailto:5...@domainb.org transfers to mailto:5...@domainb.org.
The transfer fails.
Both domainA and domainB are running on identical sipx
servers all on the same privat
n glean from a siptrace at sipx is what it sent and
> what it was sent back to make it fail. The links above provide
> ZERO meaningful insight into any call (both say request denied, from what,
> noone knows).
>
> Beyond that, a pcap from your edgmarc might prove to give other
&g
I'm experiencing some intermittent call failures and I'm having trouble
diagnosing the problem. About 50% of the time when a user dials out, they will
receive a fast busy, the other times the call goes through as expected. Once a
call is established there are times when the call will be disconne
Thank you! That was exactly what I was missing. CDR had failed to start
with an error message.
Mike Burden
[cid:image002.gif@01CCD9C3.DBDC26B0]
Lynk Systems, Inc
e-mail: m...@lynk.com<mailto:m...@lynk.com>
Phone: 616-532-4985
From: sipx-users-boun...@list.sipfound
older CDR is expected, but I'm curious what I might have done after
the initial install to cause SipXecs to stop collecting CRD info.
Apparently it was working to begin with, since I have two days of CDR data
before it stops, so I'm sure it's something I did after the initial in
Yes, I get very good support. I like to have half an idea about what I'm
doing before I call, though.
Mike Burden
[cid:image002.gif@01CCC169.FB749470]
Lynk Systems, Inc
e-mail: m...@lynk.com<mailto:m...@lynk.com>
Phone: 616-532-4985
From: sipx-users-boun...@list.sip
Guess I'll need to spend some more time reading the Ingate docs. :)
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, December 23, 2011 10:32 AM
To: Discussion list for users of sipXecs software
Subject: R
ring.
If I dial that extension directly it rings as expected, I can dial any
extension from C-SIP-Simple without a problem, and the registration on the
Registrations page looks normal.
Any idea why the behavior of the hunt group would be different than dialing the
extension directly?
Mike B
m attempting the same thing and
haven't had any luck. The best I've gotten is a 503 service
unavailable back from sipx.
Mike
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <63171>
Message-ID:
Just to update - the upgrade to 4.4 has greatly improved
presence performance. Thanks.
___
been rock solid. I
particularly like the Mediants for the ability to add
additional modules as necessary. The Mediant in our high
school has a PRI module as well as FXO ports to interface
with a paging system.
Mike
--
Mike Graham
Director of Information Technology
Hempfield
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <62994>
Message-ID:
Thanks for the replies. We'll move up the 4.4 up
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <62983>
Message-ID:
I forgot to add, no errors in sipxpresence.log.
___
sipx-users mail
k "on call" or do not light up at the appropriate
time. This is a large installation with 700+ users.
Mike
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
if it was the single digit forward time or
something else with the user that got overwritten when I
reapplied the forwarding rules. Thanks again for all of the
suggestions.
Mike
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive:
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <62834>
Message-ID:
I have. I'm seeing the error, but no useful clues.
"2011-08-23T20:30:52.333000Z":24622:JAVA:INFO:sipx.hempfi
uspect one
of my users entered a bad string into the call forward
field, but I can't find any way to determine which of my 700
users caused the problem.
Mike Graham
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.s
ilar
issues with 3.2.5 before reverting back?
Mike
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
nt that was just added is
occasionally scrolling messages about forking and invalid
RTCP packets. The other does not report this in the log.
Both occasionally show [ERROR] HugeBuffer:SelfCheck #xxx is
NOT FREE.
Mike
___
sipx-users mailing list
sipx-users@lis
en reported since the addition. The config files
look identical though. We're checking with our carrier for
differences in the way the PRIs are configured.
To be clear, the phone itself is not ringing - the phantom
ring is heard in the earpiece.
Mike
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <62189>
Message-ID:
For anyone encountering this in the future, it turns out I
received a bad batch of 450s from Polycom.
_
aiting beep, but an actual ring.
Unfortunately none of my tech staff have ever experienced
this first hand, or been able to recreate the ringing.
Just curious if anyone else has ever seen this happen
before.
Mike
--
Mike Graham
Director of Information Technology
Hempfield School District
mai
ut the text gets distorted as soon as the sip
application loads. There's a note in the firmware release that makes it sound
like this problem was fixed, but it doesn't appear to have worked.
--
Mike Graham
Director of Information Technology
Hempfield School District
_
sipx-users-boun...@list.sipfoundry.org [
> sipx-users-boun...@list.sipfoundry.org] on behalf of Mike Haun [
> haun.m...@gmail.com]
> *Sent:* Wednesday, May 18, 2011 5:35 PM
> *To:* sipx-users@list.sipfoundry.org
> *Subject:* [sipx-users] mute system prompt at end of voicemail greetin
SipX 4.2.1.018971
An inbound call is sent to an unattended extension.
After a few rings the call is then forwarded to the extension's voice
mailbox (per the call forwarding rule).
The caller then hears the personal greeting.
Then the following system prompt plays:
"When you are finished, press one
Aaron,
This could be possibly due to a bug with the UA from-Tag changing when
authenticating against sipX. Assuming the phone gets a 401 back, what are the
errors in /var/log/sipxpbx/sipregistrar.log?
Mike
Aaron Pursell esgw.org> writes:
___
s
On Fri, May 6, 2011 at 9:54 AM, Laurentiu Ceausescu wrote:
> On Fri, May 6, 2011 at 5:45 PM, Mike Kelly wrote:
>
>> Hello everyone!
>>
>> I am new to the world of sipx coming from asterisk/freepbx. I have just
>> been playing with a single standalone 4.4 server in
Can anyone point me in the right direction to look? As I said, I am
new and I am still not sure of what info gets logged into which logfile.
Thanks!
Mike Kelly
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.o
m just not
getting it?
On Wed, May 4, 2011 at 3:05 PM, Tony Graziano
wrote:
> the call is not being sent to port 5080 by the new itsp...
>
> On Wed, May 4, 2011 at 2:09 PM, Mike Haun wrote:
> > I have Sipx 4.2.1 running for a client and they've been very happy for
> about
&g
I have Sipx 4.2.1 running for a client and they've been very happy for about
a year.
I just switched sip providers from A to B. The only thing I had to change
in my sipx configs was the IP address of of the SIP Trunk gateway.
Here's the odd thing:
Test 1
A call comes in and finds it's path in a
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <58376>
Message-ID:
Has there been any resolution to this? I've had the exact
same problem and can't figure it out.
I'm running S
Now that I re-read that JIRA and understand it, I think that would be
sufficient. I think the biggest problem is when the dial-by-name switches to
list mode after one keystroke, when the user is expecting to type at least two
or three.
I logged in and added my vote.
Mike Burden
Lynk
ware
Cc: Burden, Mike
Subject: Re: [sipx-users] DIal By Name
If it makes it less complicated for the caller, why would you want to prolong
their button pushing?
On Fri, Mar 11, 2011 at 10:58 AM, Burden, Mike wrote:
> In a semi-related issue, in a very small company (like mine), it's annoy
In a semi-related issue, in a very small company (like mine), it's annoying
that the dial-by-name switches from input to selection when it has narrowed the
field to three matches. In our office, that's half the company!
Is there a setting that controls this? If not, is it JIRA-worthy?
Sorry for jumping in late, but are you sure that Gary is included in a group
that has the "List in Directory" permission?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Dan White
Sent: Thursday, February 03,
AA picks up inbound calls to this DID after 10 rings.
Is there an internal limit in sipXecs that I'm running into, or is there
another configuration that I may have missed?
Our intent for this system is that the AA never answer (at least not before a
LONG period of ringing.)
Mike B
don't have a way to make this work.
Mike Burden
[cid:image002.gif@01CBCF87.8AEACF50]
Lynk Systems, Inc
e-mail: m...@lynk.com<mailto:m...@lynk.com>
Phone: 616-532-4985
<><>___
sipx-users mailing list
sipx-users@list.si
This did not appear to have any effect. MF still starts out at 20 and
decrements to 12 after changing the forward from the extension of the hunt
group to the name of the hunt group.
Mike Burden
[cid:image002.gif@01CB9C58.36619110]
Lynk Systems, Inc
e-mail: m...@lynk.com<mailt
ou want to forward to d...@sipdomain.com<mailto:d...@sipdomain.com>
rather than forward to "300"
Thanks! I’ll try that.
And yes, the ITSP has already reconfigured to increase the Max Forwards.
Maybe I’m just being anal about trying to have the cleanest, most efficient
con
ausing two decrements,
and it appears to "bounce" 4 times before the INVITES go out to the extensions.
Our ITSP configured their switch to double the incoming MF before delivering
the call to us. Before that, we were sometimes running out of forwards.
Mike Burden
Lynk Systems, Inc
e-mail:
wards that
sipXecs consumes while bouncing the call around internally, but I'm guessing it
will be a while before that issue is addressed (and I'm not entirely sure it
CAN be addressed, if there is a possibility of a loop within the sipXecs
components.)
Mike Burden
[cid:image002.g
ds" stopped
decrementing at 12.
Mike Burden
[cid:image002.gif@01CB9B9F.ABF4BEC0]
Lynk Systems, Inc
e-mail: m...@lynk.com<mailto:m...@lynk.com>
Phone: 616-532-4985
<><>___
sipx-users mailing list
s
MPORARILY --|
|--- ACK >|
Is this normal behavior, or is it indicative of a problem in my configuration?
More details available, if they will be useful.
Mike Burden
[cid:image002.gif@01CB9B6F.840CAF10]
Lynk Systems, Inc
e-mail: m...@lynk.com<mailto:m...@lynk.com>
P
CSeq: 929 BYE
Contact: "Anonymous"
Max-Forwards: 69
User-Agent: Sippy
Cisco-guid: H323ConfID
H323-conf-id: H323ConfID
Content-Length: 0
Mike Burden
[cid:image002.gif@01CB960E.548B6080]
Lynk Systems, Inc
e-mail: m...@lynk.com<mailto:m...@lynk.com>
Phone: 616-53
r
to that Customer's CSR, or creating a "blacklist" of callers that should be
sent directly to voicemail (like the company that calls us every week asking
for the serial number of our copier. I can't BELIEVE anyone is still trying
to run that scam!)
Mike Burden
[cid:
OK, now that I've finally got sipXecs updated from 4.0.4 to 4.2.1, it sounds
like I should be updating my IP550 and IP650 phones from 3.1.3 to 3.2.1, right?
Mike Burden
[cid:image002.gif@01CB8FAB.AE265B40]
Lynk Systems, Inc
e-mail: m...@lynk.com<mailto:m...@lynk.com>
Phone: 6
Am I missing something, or do the 3.1.x and 3.2.x firmware levels not even
appear on that?
I'm really confused now!
Mike Burden
[cid:image002.gif@01CB8FAB.55FDFF40]
Lynk Systems, Inc
e-mail: m...@lynk.com<mailto:m...@lynk.com>
Phone: 616-532-4985
From: sipx
IP550 phones were nearly identical, except for
the color screen and the extra soft buttons.
Mike Burden
Lynk Systems, Inc
e-mail: m...@lynk.com
Phone: 616-532-4985
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive:
IP550 phones were nearly identical, except for
the color screen and the extra soft buttons.
Mike Burden
[cid:image002.gif@01CB8FA1.E71B1710]
Lynk Systems, Inc
e-mail: m...@lynk.com<mailto:m...@lynk.com>
Phone: 616-532-4985
<><>___
S5 system,
but now I get:
YumRepo Warning: not using ftp, http[s], or file for repos, skipping -
$releasever is not a valid release or hasnt been released yet
I'm running out of ideas about how to patch around the problem.Can anyone
tell me the RIGHT way to make yum happy again?
Thanks!
The update is scheduled for Friday at noon, so I won't know until then if
everything worked as planned.
Mike Burden
[cid:image002.gif@01CB8BCE.D4232CF0]
Lynk Systems, Inc
e-mail: m...@lynk.com<mailto:m...@lynk.com>
Phone: 616-532-4985
From: sipx
I've *FINALLY* got the go-ahead to update our sipXecs system from 4.0.4 to
current.
When I go to System --> software Updates and click "Check for Updates" I get:
No software updates available
The package information was last updated Nov 24, 2010 11:40 AM.
The system is currently running sipxcomm
have never gotten around to doing the work. That would
> reduce the consumption
> of hops in sipXecs by a factor of 2.
Sounds like a possible good start.
> Dale
Mike Burden
[cid:image002.gif@01CB7C17.7E5BF980]
Lynk Systems, Inc
Phone: 616-532-4985
www.lynk.com<http://www
Project: sipXecs
Issue Type: New Feature
Reporter: Michael W. Burden
Priority: Minor
Mike Burden
[cid:image002.gif@01CB7C11.D4D70260]
Lynk Systems, Inc
Phone: 616-532-4985
www.lynk.com<http://www.lynk.com>
From: Joegen Baclor [mailt
sipXbridge to add a few extra hops to
the incoming call to account for this?
Mike Burden
[cid:image002.gif@01CB7C05.299432D0]
Lynk Systems, Inc
Phone: 616-532-4985
www.lynk.com<http://www.lynk.com>
<><>___
sipx-users maili
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
ot, I can find the trace that the
ITSP sent me and pick the re-INVITE out of it.
Mike Burden
Lynk Systems, Inc
e-mail: m...@lynk.com
Phone: 616-532-4985
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jean-H
ked that
spells this out more clearly
or
2. Can someone tell me what specific parameters of the session are changed by
the re-INVITE when an internal transfer is performed? (I assume that said
parameters will all be in the list of things that can be changed by a re-INVITE)
Mike Burd
+1
On Wed, Sep 1, 2010 at 4:44 AM, Michal Bielicki <
michal.bieli...@seventhsignal.de> wrote:
> Douglas Hubler schrieb:
> > On Tue, Aug 31, 2010 at 10:33 PM, Flatfender
> wrote:
> >
> >> Currently our mailing lists are set to reply to the user and not the
> >> list in the mail headers. As a resu
1 - 100 of 273 matches
Mail list logo
