If the 666 (devil) is the first dial rule, and other dial strings hit it, what
causes the match? Do you have 666 as optional possibly? Look at advanced and
ensure it is not optional or anyone dialing 10 digits will hit it.
Sent from my twiddling thumbs.
Henry Dogger h.dog...@telecats.nl
What tony said. Set DND on the line on the executive phone - it simply
turns off the ringer on it, or set it to a ring tone that is silent. Create
a secondary extension that is in an odd range, and then create a dial plan
that excludes that range, and a dial plan that includes it. Assigned the
I'm not sure your question is clear. Are you asking to have separate MOH
for auto attendant transfers, versus music that is heard when placed on hold
by an individual?
If that is your question, I've not seen a method that is designed
specifically to provide one type of Music for Auto
Intercom for a Softphone? I don't think you will find that supported.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Roman Gelfand
Sent: Wednesday, January 02, 2013 3:00 AM
To: George Niculae
Cc: Discussion
Of Roman Gelfand
Sent: Wednesday, January 02, 2013 9:38 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Intercom in 4.6
This worked in 4.4 for all phones. Is this a security feature? Is there a
way to include all phones?
On Wed, Jan 2, 2013 at 12:06 PM, Todd
I handled this by creating multiple FTP usernames - each site has its own
username. On the FTP site, the FTP user has their own directory. This
created a group of directories, each containing its own backups - and each
directory has a sub-directory by date with the backup files in it.
Very
Douglas - that is a nice improvement for the FTP. :) As always - Thanks!
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Friday, December 21, 2012 1:50 PM
To: sipx-users@list.sipfoundry.org
After you reset the Bridge, and until it is fully reset, you typically see
this error. Is this machine real slow possibly?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matt Nelson
Sent: Tuesday, December 18, 2012 7:30 PM
To:
You can run log rotate to remove the log files and monitor smaller one.
logrotate -f /etc/logrotate.d/sipxchange
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Sunday, December 16, 2012 8:58 AM
This is kind of a useless email to send out to the mailing list. Vent, but
personally, I'd prefer my email box not get filled with it. This release
has been out there for the most part for about 6 months. Everyone has had
the same opportunity to load it on a test system, and test their own
Maybe a stretch - Checked the POE switch? Had a phone resetting just the
other day when they hit the messages button on the phone - as bizarre as
that sounds, it turned out to be a cable issue between switch and phone. Go
figure?!? Still scratching my head.
-Original Message-
From:
has done this already, I would love to hear details on it.
Thanks in advance for any information you might have.
Regards,
Todd Hodgen
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive
Congratulations Douglas to you and your entire Team. This substantial
release demonstrates a new level of flexibility and reliability to an
already world class product! A warm thank you to everyone involved!!!
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
Nicholas - Your details are pretty extensive here, so I may be pointing out
the obvious - but have you confirmed what ports the RTP is using, and if it
is configured in the router to be passed. Not always, but many times I
find RTP issues are generally related to ports configuration (or not) in
You can also use a low cost ATA (Linksys, Grandstream, Audiocodes) that
allows you to call an extension number, the ATA pots port connects to a
valcom or bogen Page Adapter, and any number of standard speakers. Old
fashioned way of doing things, but it works very reliably.
All three of those
That should work. Alternatively, you can put the did for the fax as the fax
extension number - just one number to deal with then.
When you call the number from another phone, do you hear Fax tones?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
Or, User is 12345, and unified messaging has a fax mailbox of
2223334567
In maillog, for the fax call, what DSN does it provide for the mail forward?
2.0 or something else?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
You should see something from mail going out. So you are broken somewhere
ahead of that. Are you hearing fax tone when you call it?
Normally, when things are broke, you at least get an email that says 0 pages
received. At least that has been my experience.
-Original Message-
From:
I'm not sure just how important the Bria Softphone really is. Personally, I
don't use it or recommend it to customer. Many have found their support
lacking, and they don't seem to be Channel friendly.
Have you tried other clients? I've had good results with the 3CX softphone,
and they
I suspect your issue is with Verizon, and not the router. They are
blocking your ability to send email through them as an SMTP gateway.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent:
Specifically, we do not use VOIP.ms as primary service for customers. We
find they are not reliable enough. They are great for testing, quick
temporary service, occasional usage such as conference bridge, international
dialing, etc.
I don't believe their issues are related to routers in most
Yes, the Seattle switch is horrible, and I don't recommend it. They say
that Seattle and Chicago are their two newest and most up to date switches,
as well as least busy. But I can't use Seattle any longer.
From: sipx-users-boun...@list.sipfoundry.org
Have you consided using a simple low cost Polycom phone along with Voice
Operator Panel as a solution? It works great!
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Geoff Musgrave
Sent: Tuesday, November 20, 2012 11:11 AM
To:
You could set up multiple extensions with call forwarding to a particular
number. Ext 111 - 206-4567890, Ext 112 - 206-2254589, etc. From auto
attendant, dial the extension.
If you explain what you are trying to do in more detail from an application
standpoint, I suspect there are other
Under line - Proxy and Registration - be sure to check Use Outbound Proxy
and enter an outbound Proxy if not already checked.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of ??
Sent: Friday, November 16, 2012
it's a public IP or not.
~Noah
On Nov 15, 2012, at 7:07 PM, Todd Hodgen thod...@frontier.com wrote:
Here is a question I would have as well - 172.129.67.195 seems to be an
address that is local to your network. Who has that IP address, why are
they attempting to breach that server
Look at var/spool/mail/rootThere is a report you can find in there that
shows system activity. Look for entries below -
pam_unix Begin and I think you will find the source
of your aggravation.
-Original Message-
From:
+1
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Thursday, November 15, 2012 7:49 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Hacked SipXecs 4.4
yes, and using the word
On Nov 15, 2012, at 12:42 PM, Todd Hodgen thod...@frontier.com
wrote:
+1
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org]On Behalf Of Michael Picher
Sent: Thursday, November 15, 2012 7:49 AM
To: Discussion list for users of sipXecs software
Subject
, as every sipxecs install probably has this user with a
default password?
~Noah
On Nov 15, 2012, at 12:41 PM, Todd Hodgen thod...@frontier.com wrote:
Look at var/spool/mail/rootThere is a report you can find in there
that
shows system activity. Look for entries below
Shane, Your efforts to continue to use the Asterisk as an ATA, although
valiant, might be a daunting task. Have you considered simply picking up a
used Linksys ATA on ebay for $20-$30 and calling it a day? I suspect their
might be some resell value in your current IP04 to help pay for it. I
There is a sample provided in the like I had sent. Copy and paste it,
that's all you really need to do, and then change the from field to your
own.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
The SPA 504 templates were donated by two developers, and introduced in 4.4
A later email I had read from one of the developers indicated there were
issues, and they were not going to address them because each Cisco release
made changes that created a moving target.
I do have a customer that
I've not been following this thread, but will add some comments as they
relate to email not being deliverable outbound via sipxecs. If this is DNS
related - this response doesn't touch that.
I find it useful in v4.4 to add the Emailformats.properties files to
sipxecs.
You can always create random generated passwords to fill in your spreadsheet
as well using a formula.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Brent Besse
Sent: Thursday, November 01, 2012 11:10 PM
To: Discussion list for users
I had a server that needed to be rebuilt yesterday with 4.4. I ran a backup
of voicemail and configuration, rebuilt the server, but had an error when
trying to restore the voicemail.Don't have exact error, but it was
complaining the file was too large. And it was big, about 1.2 tb.
should be 0644.
On Tue, Oct 30, 2012 at 2:28 PM, Todd Hodgen thod...@frontier.com wrote:
I had a server that needed to be rebuilt yesterday with 4.4. I ran a backup
of voicemail and configuration, rebuilt the server, but had an error when
trying to restore the voicemail.Don't have exact
Henry, Try allowing the existing sipxecs template configure the phone,
without making the changes in the wiki to the profiles.
I have a system with approximately 20 of those phones working perfectly with
the system managing the templates for the phones completely.
From:
is the difference in handling transfer to voice mail between an internal
and external call?
I apologize for all these questions but I just am mystified by my encounters
and observations.
Thanks and best regards,
Henry Kwan
From: Todd Hodgen thod...@frontier.com
To: 'Henry Kwan' hslk
I don't think this jells completely with your description, but when you do a
restart of a service, it does take some time for the restart and the
services to work again. The sipxbridge is a good example, because it
restarts, and then must re-register before you can make call on it.
Check on
of sipXecs software
Subject: Re: [sipx-users] 4.6: polycom and BLF
How can you have BLF on a speed dial button that is programmed as *76750?
Why don't you add line 750 to the phone, have it silent ring and try to pick
it up that way (BLA)?
Mike
On Mon, Oct 15, 2012 at 10:19 PM, Todd Hodgen
I have 4.6 running with Polycom 550 and Polycom VVX500 and show presence on
each phone when the other is used.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Sunday, October 14, 2012 10:52 PM
To: Kumaran
Cc:
George, Wiki still seems to be down. Do you have access to service it?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Sunday, October 14, 2012 10:52 PM
To: Kumaran
Cc: Discussion list for users of sipXecs software
Explain the scenario. Call ringing on a phone, and you want to do a
directed call pickup from a button that has a speeddial button configured?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel
Watkins
Sent: Monday, October 15,
1500 doesn't see the status of either extension - pressing the speed
dial button of the extension being called simply initiates another call.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: Monday, October 15, 2012 4:51
Henry, I can't speak to the router, or your ITSP provider. I can state
that I have a site running on 4.4 with a single server, server provides DHCP
and DNS, and works with SPA942 phones. I did not use the wiki
recommendations. I simply provisioned them via the management templates and
they
In the past, I've used a plugin to provide a User Summary Page for sipxecs.
It is needing to be rewritten, and I'm looking for comments as to whether it
should be included in the core software for all to have access to, or as a
plugin that would only be available if you drop it into your
In my testing, it was during the playing of the mailbox message. Pressing
the * should allow you to move to another voicemail box by entering the
extension and password of an alternative box. In my scenario, it terminated
the call.
-Original Message-
From:
for users of sipXecs software
Subject: Re: [sipx-users] 4.6 voicemail hangup when press *
On Fri, Oct 5, 2012 at 9:26 AM, Todd Hodgen thod...@frontier.com wrote:
In my testing, it was during the playing of the mailbox message.
Pressing the * should allow you to move to another voicemail box
I was trying to get to my voicemail from another phone to test something and
when I pressed *, rather than allowing me to enter my alternative extension
and password it hung up on me. Using Polycom 550 and vvx500 same results.
I'd like to confirm if others see the same thing on 4.6?
What
...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Thursday, October 04, 2012 11:06 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6 voicemail hangup when press *
On Thu, Oct 4, 2012 at 8:16 PM, Todd Hodgen thod
when press *
On Thu, Oct 4, 2012 at 10:18 PM, George Niculae geo...@ezuce.com wrote:
On Thu, Oct 4, 2012 at 10:01 PM, Todd Hodgen thod...@frontier.com wrote:
Hi George, thanks for the comments. Not related to MWI.
Scenario - from extension 701 - dialed 8750 to go directly to ext 750
-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Saturday, September 29, 2012 3:12 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Lastest Patches and Alias
Yes, just sipxregistry and sipxconfig rpms
George
On Sep 30, 2012 1:03 AM, Todd Hodgen thod
] On Behalf Of Todd Hodgen
Sent: Monday, October 01, 2012 5:47 PM
To: 'Discussion list for users of sipXecs software'
Subject: Re: [sipx-users] Lastest Patches and Alias
George, seeing an error - Config error: file contains no section headers
File: file:/etc/yum.repos.d/sipxecs.repo
file:///\\etc
I haven't dug into it yet, but will in morning. Last night I did a Yum
Update to a site that was running on 4.4, with a Patton Gateway to PRI.
Everything was running well, but the certificate was about to expire.
Here are the steps I took -
Stopped sipxecs
Yum Update to get the latest
...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Saturday, September 29, 2012 3:43 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Lastest Patches and Alias
On Sat, Sep 29, 2012 at 1:33 PM, Todd Hodgen thod...@frontier.com wrote:
I haven't dug into it yet, but will in morning
-users] Lastest Patches and Alias
Todd,
you should yum update sipxregistry and sipxconfig from here for the moment,
they contain changes reverted:
http://download.sipfoundry.org/pub/sipXecs-stage/4.4.0/CentOS_5/x86_64/
George
On Sat, Sep 29, 2012 at 10:37 PM, Todd Hodgen thod...@frontier.com
, Todd Hodgen thod...@frontier.com wrote:
Sorry I don't.
This is unfortunate, as it creates service issues for this site.
This brings up a concern with this patching process that is being used.
Yum update pulls in all of the patches, and has the potential of
bringing in patches
of sipXecs software
Subject: Re: [sipx-users] Lastest Patches and Alias
Ah, you should try from here
http://download.sipfoundry.org/pub/sipXecs-stage/4.4.0/CentOS_5/i386/
George
On Sep 30, 2012 12:18 AM, Todd Hodgen thod...@frontier.com wrote:
This might be a problem, as I believe the site
In the past I've removed that. I think its a separate wave at the end or there
was an alternative one to use in the directory. I didn't have to record it
Sent from my twiddling thumbs.
niklas rehnberg niklas.rehnb...@gmail.com wrote:
___
sipx-users
I've got about 20 VVX500 phones working with Music On Hold. It only worked
when the phone was provisioned via TFTP, and not manually. We never found a
method for putting the MOH server into the phone configuration from its gui.
Here is the line we use for Music on hold for line 1 -
If you install with ISO, it will find the one NIC you have attached to the
network, which you will assign an IP to. That is all there is to it.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent:
Jeff, I think the confusion is that some sip components are on the
192.168.53.x network, while others are on 172.21.201.x network. In your
description below, you state which routes, which is indicating there is a
router on the network, and the IP addressing would indicate they are on
different
Have you tried removing all the phones but one on that called user to ensure
its not a bad behaving endpoint?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jeff Pyle
Sent: Wednesday, September 19, 2012 7:16 AM
To: Discussion list for
Not exactly a regression test. Might want to do with abundance of caution
Sent from my twiddling thumbs.
Douglas Hubler dhub...@ezuce.com wrote:
This made my day,
A.) FS 1.2.1 works
B.) FS from their repo just works
Thanks Niek. I guess the answer is that is unofficially works, and
use this
Since this is an open source project, support for new phones is generally
provided by those that use them. Personally, this is the first time I've
seen this phone come up in conversation on the list, so I suspect nothing is
in the works. You may be able to manually configure the phone to work
!
~~
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013!
On Sep 9, 2012 5:21 PM, Noah Mehl n...@tritonlimited.com wrote:
I would also be extremely interested if anyone has any form of speech to
text working.
~Noah
On Sep 7, 2012, at 2:12 PM, Todd Hodgen
It's been a while since I've seen this subject on the mailing list, so
thought I would shout this out to the list. Is anyone currently using a
speech to text application with good results for the unified messages from
sipxecs?
___
sipx-users
You probably don't want a thousands Got it, but I'll give you at least
one!
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
Sent: Tuesday, September 04, 2012 6:44 PM
To: sipx-users
Subject:
I have a scenario that works with one ITSP, and not with another.
With Broadvox Fusion network, call comes into office, is answered by
receptionist and transferred successfully. Call is picked up on transfer
and placed on hold, then picked up and transferred to another extension.
I have a scenario that works with one ITSP, and not with another.
With Broadvox Fusion network, call comes into office, is answered by
receptionist and transferred successfully. Call is picked up on transfer
and placed on hold, then picked up and transferred to another extension.
Are attachments going through?
I see my response to your email went through. However, one that I had sent
that has a small attachment hasn't yet. It was sent prior to my response to
your test message.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
again until you restart ACDServer.
- sipxacd.log is full of
2012-08-30T18:44:02.139782Z:934039:KERNEL:NOTICE:sip.example.com:MpMedia:B
6BB1B90:sipxacd:OsMsgQShared::doSendCore message queue
'mpStartUp::MpMisc.pSpkQ' is over half full - count = 12, max = 14
+1 on what Todd Hodgen said about
Matt - thanks for this detailed response. Can you share with the list if
there are any gotcha's that you know of that you avoid so that you have
successful deployments of the ACD, or any best practices you have used to
ensure success?
From: sipx-users-boun...@list.sipfoundry.org
Here is a possible workaround beyond additional extensions on the phones.
Create an auto attendant that has options/replay count and options/Invalid
Response Count disabled. Check the tick for transfer on failure and set
the transfer to the hunt group that contains the extension you want to
You can try to insert a phantom extension in between your steps, that
forwards to the next step, rather than having one phantom do all of this.
Haven't tried it, but would if I was fighting this issue.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org]
Running a site on 4.4 with an issue on alarm notification. I have built
one group called 911Notify that contains three SMS and one email address for
notification when 911 is called. In the alarm group for 911, I have
selected 911Notify as the group to notify on that particular alarm.
:10 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Alarms troubleshooting
I would check the user:group and permissions for that file.
-rw-r--r-- 1 root sipxchange 10920 Jul 9 18:45 sipXalarms-config.xml
On Thu, Aug 16, 2012 at 1:16 PM, Todd Hodgen
, Aug 16, 2012 at 4:10 PM, Todd Hodgen thod...@frontier.com wrote:
Just to confirm - I am seeing this on two different machines.
Disk space used about 10% on this machine.
Directory - drwxr-xr-x 2 root root4096 Aug 6 19:17 alarms
Files in the directory -
-rw-r--r-- 1
BTW, Thanks Tony!
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: Thursday, August 16, 2012 1:52 PM
To: 'Discussion list for users of sipXecs software'
Subject: Re: [sipx-users] Alarms troubleshooting
That did
://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013!
On Aug 16, 2012 5:01 PM, Todd Hodgen thod...@frontier.com wrote:
BTW, Thanks Tony!
From: sipx-users-boun
Is this with a particular ITSP by chance?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ly Tran
Sent: Tuesday, August 14, 2012 2:23 PM
To: 'jmicc...@redhat.com'; 'Discussion list for users of sipXecs software'
4.6 is not a stable product release yet. It would seem that if you want to
use the old legacy ACD, you should remain on 4.4. It's a stable release.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas
Is there a strategy for those that use the ACD as far as migration from
Legacy ACD to OpenACD Call Center? I don't use it, but it seems that going
from 4.4 using ACD, to 4.6 with Call Center, there are some holes with
regards to agent login-logoff, reports, etc.
Maybe the development
: [sipx-users] ACD (Legacy) not working in 4.6
On Sun, Aug 12, 2012 at 7:06 PM, Todd Hodgen thod...@frontier.com wrote:
Is there a strategy for those that use the ACD as far as migration from
Legacy ACD to OpenACD Call Center? I don't use it, but it seems that going
from 4.4 using ACD, to 4.6
A few things that I've found with Sendmail, it's usually the receiving SMTP
gateway that is not happy with the domain used, or the email account.
Sometimes, adding the Emailformats file into etc/sipxpbx/sipivr and editing
it with an acceptable email address and domain resolves some of these
You have to run the 4.0.2 firmware, as there was a bug 4.0.1 that affected
these phones working with sipxecs.
Some features will only work if you provision the phone via TFTP, and not
via the GUI. Music-on-hold is one of those features.
What I've done is configured the .cfg
Interesting, on some Linksys SPA942 I had music on hold working on
phone-to-phone, and phone to gateway (Epygi). However, can not get it
working with Phone-to-Audiocodes.It did play Audiocodes noise making
Music-on-hold tone though - but it was a bit obnoxious. Not sure if they
operate the
4.4 is based on Centos 5.x, 4.6 is based on Centos 6.x. I don't believe
there is an upgrade path now, or planned, other than configuration's being
able to be moved.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
m...@grounded.net
What version? 4.4 definitely works.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Robert B
Sent: Monday, August 06, 2012 11:01 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Customizations to the
I suspect a template created by a user of aastra has a commit with an issue.
Open source. Someone using aastra will need to resolve how that template
writes its variable. Should be easy enough to resolve.
Sent from my twiddling thumbs.
Mark Dutton repl...@datamerge.com.au wrote:
Firstly,
Often times I put my extra DID numbers under either an Auto Attendant, or a
User, with a recording announcing it is a non-working number at the customer
site. This allows for people dialing disconnected numbers in the office a
means of getting to an operator, another department, another person
May have missed this detail in your earlier email - did you perform a Yum
update on the last ISO install? There are about 17 patch released to 4.4
that are not included in the ISO.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
I believe the rule based password is not a bad idea. I don't believe you
want a system configured with a rule base password, EXCEPT, at startup. If
you are rolling out a system, you need a method to train end users, and a
method of having them go back to their desk and log onto their new
really link to anything but itself.
On Thu, Jul 26, 2012 at 12:05 PM, Todd Hodgen thod...@frontier.com wrote:
The biggest selling point of sipxecs is that it uses Open Standards.
Lync does not.
Audiocodes and the Siperator both have a claim to do Lync integration as a
mediating server between
Nathaniel, Can you post a sanitized INI file of the Audiocodes for review,
or send it privately?
This application definitely does work.
A trace of a failed call would definitely help.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
There is another method of controlling this a well. With your ITSP, you can
control the number of Concurrent Call Sessions they allow. Most charge for
CCS, although some don't. This won't take into account remote workers, etc.
but that can be easily handled by an SBC.
-Original
Ah, but that is why we have Mike. Those complicated systems need a top
notch Engineer, Author and just all around fast driver to show them how to
correctly engineer the goezinda' and goezoutta' of the network. What's
that line - we don't need no stinkin' CAC, we got Mike to watch our BACK.
I
The biggest selling point of sipxecs is that it uses Open Standards. Lync
does not.
Audiocodes and the Siperator both have a claim to do Lync integration as a
mediating server between a PBX and Lync. Probably a great place to start.
Of course with Karoo now being open source, it might be a
Nathan, I'm interested in the failures you are seeing on these. I've
installed many without any failures yet. Is it a particular component that
is failing on them? Has Audiocodes engineering been brought into a
discussion on it?
From: sipx-users-boun...@list.sipfoundry.org
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