Sounds like an order for more Polycoms!
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh M. Patten
Sent: Thursday, November 04, 2010 5:55 PM
To: Discussion list for users of sipXecs software
Subject: [sipx-users] Look what happens wh
I am able to confirm an attended transfer to media services works now.
I can also dial a users voicemail box with the default of "8+sipxuser".
I am finding I cannot do a blind transfer to "8+sipxuser" to send the caller
directly to voicemail from a phone. Is it just me (I am on the latest
snapsho
I can't laugh. I've seen this type of thing before. Since it could make
calls you would think the telco stuff was fine (it probably was). I have
heard of big fancy phone switches hiccupping before. As far as the audio
interview goes, I'd never leave it open as "we can't really say at this
moment",
On 11/4/10 8:54 PM, Josh M. Patten wrote:
http://www.wtaw.com/2010/11/04/no-incoming-calls-to-brazos-county-courthouse/
good thing we never have to reboot sipx to get service, or clear our
java memory overload.
It turns into a PR nightmare (running Nortel Option 61c installed in
1999. Runnin
http://www.wtaw.com/2010/11/04/no-incoming-calls-to-brazos-county-courthouse/
It turns into a PR nightmare (running Nortel Option 61c installed in 1999.
Running last software rev that supported Meridian Mail).
I wonder what the elected officials are going to do tomorrow...
__
> Why are there 2 Gateways in the Network of your sipx? Which nets are they
> connected to?
I have several gateways on the network, each for it's own traffic to prevent
packets flowing on to network segments they don't belong on. It's also a
security measure of course.
> If there is only one sp
On Thu, Nov 4, 2010 at 8:08 PM, Dan White wrote:
>
> This only works in older versions
I'm pretty sure this should work in latest versions too. What kind of
issues do you hit? Or provide files and I'll take a look at.
George
___
sipx-users mailing list
To Access the Audio Conference Bridge for tomorrow's Wiki clean-up efforts:
ISN #:
32008*1312
Sip URI#:
32...@openuc.ezuce.com
PSTN:
978-296-1009
To enable ISN Dialing in sipXecs:
System -> Servers
Services -> Sip Registrar
Check 'ISN Dialing'
...
-Original Message-
From: sipx-user
Kyle - you are correct - we had to update the war files as well.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kyle Haefner
Sent: Thursday, November 04, 2010 3:16 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-us
I've always have had better look un-waring the sipxconfig.war file changing
files (like the logo) and re-warring the file and restarting sipx services,
it's a pain because it doesn't survive upgrades. Too bad it doesn't work
like themes in Joomla or drupal, now that would be slick!
Kyle
On Thu,
I think what Tony is saying is - he thinks the problem is with your ITSP...You
know Tony - mr. beat around the bush...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, November 04,
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This only works in older versions
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sipx-
No there are not symlinks.
Get thee to a reputable ITSP.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Cont
yes, even now when I view the sipxbridge.xml is showing "loose
routing" tag as false.
I just realized something... is there symbolics links to these files?
If yes, from where?
On Thu, Nov 4, 2010 at 1:29 PM, Tony Graziano
wrote:
> No.
>
> Verify the edit stayed in place after saving it.
>
No.
Verify the edit stayed in place after saving it.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract
That is just the thing, the sipxbridge.xml is not getting overwritten
and yet the sipxbridge.log is showing outgoing requests with ;lr
tagged to sip uri. I had rebooted the machine.
Perhaps, the xml files are being merged some place and files got locked?
On Thu, Nov 4, 2010 at 1:08 PM, Tony Gr
On Tue, Nov 2, 2010 at 6:39 AM, Abdul Mayat wrote:
> I think the answer is no, but i though I would double
> check!
>
> Is it possible to localize the IVR (specifically voicemail)
> by group (or any other parameter) so the voice prompts
> played out are specific to the users language?
>
> Currentl
This is why you should not use an ITSP who is nowhere near RFC3261
compliant.
Edit sipxbridge manually and just "restart" the sipxbridge service.
As I stated, any changes to your profiles that would project them will
overwrite it.
Consider a more compliant ITSP.
Tony
For weeks now my sipx configuration has been working great. The
reason why it worked well is because I set "loose routing" tag to
false in sipxbridge.xml. So the sip uri didn't have ;lr. Now, no
matter what I do I can't get ;lr out.
Any help is appreciated.
Thanks
_
this has been discussed before.
1. 4.2.0 is a bit buggy and going to 4.2.1 is a good idea, but be sure you
follow the correct path and get to the right 4.2.1. See the wiki.
2. It is likely you need to kill the config server and relaunch is.
sipxproc --help
(sipxproc -r ConfigServer) to restart
I've come to the same solution, that counterpath could do a better job
before releasing in making sure things work as expected. I won't go down
that road here...
I have noticed that the support has been "lackluster", I spend 20-30 hours
troubleshooting their prouduct, which I paid 60.00 for, yet s
On Thu, Nov 4, 2010 at 11:47 AM, Dan White wrote:
> In version 42 of sipx, can you change the web front end to
> your colors, logo, and put links on the page?
>
> you were able to do this before, but I can not seem to find
> any of the pages that I need to change.
http://wiki.sipfoundry.org/displ
I'll get in the #sipx IRC channel on freenode.net around 3AM EDT and
kick off the event with details. Don't worry if you miss it, I'll
help people get started no matter when they join.
NOTE: 3AM EDT is GMT +4 or 7 AM GMT
Things you can do to prepare:
1.) If you plan on making a fairly signif
Hi,
We have multiple setups with sipXecs 4.2.x and are facing some problems with an
instance installed on a vmware machine.
Version 4.2.0 is installed on this machine and from time to time the
webinterface starts getting slower, and in the end it will stop working at all.
We are only facing
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Worley, Dale R
(Dale)
Sent: Thursday, November 04, 2010 11:45 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Too many hops?
[.
Hi Paul,
I think the most frustrating part is that everything worked in Bria 2.5 and
now several things do not. The things I like the least:
1. It requires a new C++ runtime to be installed, with windows XP computers
and certain hot fixes Bria would always crash on startup, the workaround is
t
It's not a mystery. The client user agent for RLS is set to grab a
random port.
mClientUserAgent(
PORT_DEFAULT, // sipTcpPort
PORT_DEFAULT, // sipUdpPort
PORT_DEFAULT, // sipTlsPort
NULL, // publicAddress
NULL, // defaultUser
bindIp, // defaultSipAddress
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Burden, Mike
[...@lynk.com]
The ITSP says that the calls arrive at their switch that way.
Looking at the sipxtrace, it looks like the modular architecture
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Organization: SipXecs Forum
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Message-ID:
In version 42 of sipx, can you change the web front end to
your colors, logo, and put links on the page?
you were able to d
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;) Bump!! :d
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Thanks. I took your suggestion.
Reduce effect of the modular architecture on the hops consumed by sipXecs
-
Key: XX-9190
URL: http://track.sipfoundry.org/browse/XX-9190
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
http://track.sipfoundry.org/browse/XX-9189
I'm mystified why this just started happening - any ideas? And while I
have a workaround, it is quite crippling to my user base to restrict the
RTP range.
joe
On 11/04/2010 10:41 AM, Joegen Baclor wrote:
>
Some calls are coming into sipXecs with a low number (5 of less,
sometimes 3 or less) hops remaining.
The ITSP says that the calls arrive at their switch that way.
Looking at the sipxtrace, it looks like the modular architecture of
sipXecs "eats" a lot of hops (ITSP to sipXbridge, sipXbridge t
I don't think this is configurable now. However, I think this is a
good point. Why don't you create a jira feature request perhaps we
can catch it one of the future iterations.
On Thursday, 04 November, 2010 09:46 PM, Burden, Mike wrote:
Some
calls are coming into sipXecs w
Hey Joe,
Go ahead and create a jira ticket. We've received some info from Ranga
that this might have some undesired complications in other parts of the
code. We will investigate further and update the issue tracker accordingly.
Joegen
On Thursday, 04 November, 2010 09:49 PM, Joe Micciche wro
The soundfiles yes
the rest YMMV
Am 04.11.2010 um 02:28 schrieb Sven Evensen:
> I found it googling. I eventually found the link you mention too, but as
> you mention, it is only English. I find a number of languages for 4.0,
> can they be used for 4.2 sipx?
>
> Sven
>
> -Original Message--
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Joegen, do you want me to open a case for this or have you already?
> This is a one liner fix in the relay code.
>
> for (int i = config.getPortRangeLowerBound(); i < config
> .getPortRangeUpperBound(); i++) {
> try {
>
Some calls are coming into sipXecs with a low number (5 of less, sometimes 3 or
less) hops remaining.
The ITSP says that the calls arrive at their switch that way.
Looking at the sipxtrace, it looks like the modular architecture of sipXecs
"eats" a lot of hops (ITSP to sipXbridge, sipXbridge to
I don't see what's problematic if there is no problem in the first place.
First, spark does support SRV's, so no preblem there.
Second, according to Douglas hostname needs to be XMPP domain name, so I
started to verify this.
I can't find the XMPP-domain==hostname relation.
I did the following
It's more problematic if this is going to be the standard way they do things
going forward. It more likely means that using DNS to overcome it is the
easier solution if the openfire folks are not going to accept a change. This
would mean the superadmin would need to be able to maintain some A recor
I traced and see no problemo with Spark.
Attached a wireshark with filter "port 53" during the start of
spark2.6.0b2 after I did an c:>ipconfig /flushdns.
Unfortunately my SRV and A record are the same, but that's normal when
you're a Bria user :o)
Paul
Douglas Hubler wrote:
> On Wed, Nov
Kyle,
If you would describe what is not working for you then people might come
up with a solution.
Also describe what features you need.
If you only need audio then you can easily find (non-provisionable)
alternatives.
Bria 3.1.2 is actually acceptable IMHO (I won't say it is good).
The provis
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