Try under System/Server/SIP Trunking/ SIP There is a tickmark to provide
Music on Hold. Default is ON, so it should be working unless it has been
specifically turned off.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On
Thanks. Misiu Systems funded that development, and a big thanks to George
for finishing off the final touches on it.
BTW, not to hijack the thread, but that process showed me how well
collaboration works with the Community and the Developers. We funded the
nuts and bolts of it, the
.
On Tue, Mar 20, 2012 at 3:19 PM, Todd Hodgen thod...@frontier.com wrote:
Thanks. Misiu Systems funded that development, and a big thanks to George
for finishing off the final touches on it.
BTW, not to hijack the thread, but that process showed me how well
collaboration works
Might try running 5.6 or 5.8 code to see if that helps. No reason for it,
but I haven't been running the 6.x code. Mainly because I haven't seen
anyone state they are working with no issues on it, and I've seen that on
the 5.6 and 5.8 code.
From: sipx-users-boun...@list.sipfoundry.org
to do so.
On Mon, Mar 19, 2012 at 1:55 AM, Todd Hodgen thod...@frontier.com wrote:
Has anyone done testing with Airespring SIP trunks recently with sipXecs?
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List Archive: http
Robert, 3CX has a free softphone that can be downloaded from their website
at 3cx.com
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Robert
Schroeder
Sent: Monday, March 19, 2012 1:50 PM
To: sipx-users@list.sipfoundry.org
Subject:
The great thing about Open Source is the Source is Open, and features can be
developed by anyone. I'm sure others would welcome the contribution.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Emilio
Panighetti
Has anyone written a CDR Plugin for Sipxecs? I seem to recall some
discussions around it, and the API.
Or, is anyone using an existing Call Accounting Application with sipXecs
with success?
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Have you considered a UPS product that has orderly shutdown capability?
There are some that will do controlled shutdown and startup of select
devices and in your selected order. You could have it turn up your DNS
services first, with sipXecs system coming back online after a selected
timeframe,
] Softphone for Apple
Bria for Mac works well including jabber support.
On 03/15/2012 04:23 PM, Michael Picher wrote:
I haven't tested either but Counterpath Bria is available for Mac as is
Jitsi...
Mike
On Wed, Mar 14, 2012 at 11:39 PM, Todd Hodgen thod...@frontier.com wrote:
Has anyone had
Has anyone had good luck with a softphone for Apple that they can recommend?
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Joe, I think if you enable t.38 in your gateway, things will begin to work.
I have customers receiving calls via PSTN connected fax machines, PRI
gateway has T.38 enabled, they are forwarded to customer voicemail account.
I did have to go into the gateway and turn on T.38 for it to start working.
, only t.38. T.38 is a requirement.
On Mar 13, 2012 4:57 PM, Todd Hodgen thod...@frontier.com wrote:
Joe, I think if you enable t.38 in your gateway, things will begin to work.
I have customers receiving calls via PSTN connected fax machines, PRI
gateway has T.38 enabled, they are forwarded
Good Call!
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Mark A. Smith
Sent: Sunday, March 11, 2012 12:07 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Sending Hook-Flash to FXO Gateway
Mike, you had a typo…… Weren’t you thinking 15.2?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Thursday, March 08, 2012 4:31 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] CoLab
: [sipx-users] CoLab
On Thu, Mar 8, 2012 at 1:15 AM, Todd Hodgen thod...@frontier.com wrote:
Douglas - will there be any published results for Hackfest?
A lot of the guys are visiting US for a while, but I can give you some info
i gathered. This is all work that will go into 4.6
Da Gangstas
Can you determine the type of fax they are coming from? If you have control
of that fax, for troubleshooting, turn off ECM and slow it down to 9600 and
test again.
When using T.38, you generally need to disable Error Correction Mode. A
major symptom of ECM being used is you only receive the
Darn, I really regret missing CoLab, sounds like it would have been a great
event.
Okay, Kidding. It was a great event, well organized, and a great
opportunity to meet like-minded Open Source advocates. If you missed it, I
do recommend it for next year.
It was great meeting all of
Sorry, Thanks to Dave and Joanne, I put the wrong names in. My half-timers
kicking in!
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: Wednesday, March 07, 2012 6:52 PM
To: 'Discussion list for users of sipXecs
Actually, the book is called “The Book formerly called Building Enterprise
Ready Telephony Systems”.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel Watkins
Sent: Wednesday, March 07, 2012 8:34 PM
To: Discussion list for
...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: Wednesday, March 07, 2012 11:37 PM
To: 'Discussion list for users of sipXecs software'
Subject: Re: [sipx-users] CoLab
Actually, the book is called The Book formerly called Building Enterprise
Not a Broadvox Guru for sure, but I've had some good luck with them.
Actually, their new fusion platform works fine now also. You have to
request the Mitel Option, and ask that they configure your contact in the To
field, and everything works great.
I've worked with Steve a bit, but I'm
Are they really sending NI2? I'd confirm with them that they are NI2. I
had a similar issue, with a different brand of Gateway, and it turned out
they weren't sending NI2. Everything worked, but the Name. Once we lined
up to the same ISDN protocol, it worked great.
From:
Look at the logfiles. I had a similar issue with a VVX1500 just last week.
Phone was completely messed up. Log indicated it couldn't load the
firmware, the error was full disk on the phone (huh?). I updated the
bootrom and everything worked fine after that.
I think there was an old bootrom on
Ingate has just announced a software only version of their product, which
also includes Intrusion detection and Intrusion protection. Might be
promising and a worthwhile plugin to develop.
http://www.ingate.com/2012-02-01%20Ingate%20Systems%20to%20Offer%20New%20Sof
tware-Only%20E-SBC.php
http://www.centos.org/docs/5/html/Deployment_Guide-en-US/ch-swapspace.html
If you have adequate memory, services should not be using Swap, since swap
is the use of hard disk space for when memory is short for swapping
applications in and out.
From: sipx-users-boun...@list.sipfoundry.org
If you upload the desired Polycom files, it will overwrite the files in most
cases.
Here is the location of the TFTP files -
var/sipxdata/configserver/phone/profile/tftproot
The Polycom files, and the [mac]*.* files will be related to your phone.
I'd be careful, and copy those files to a
If you are using an older Polycom phone, like the 301, 501, 601 or even
older, open the .sip file and look at the file they are looking
for, as it is an older version of the firmware. You will see a reference to
the model of phone, and the exact LD file it is requiring, and it will be
Hey Tony, Didn't want to highjack Dave's thread, so I have renamed it and
kept the past emails in it. You have gotten my interest here with your
last response - have you proven that the issue with parked calls not being
returned was related to the Patton Gateway? I've seen that issue, but had
and is not scheduled
yet.
On Feb 24, 2012 7:25 PM, Todd Hodgen thod...@frontier.com wrote:
Hey Tony, Didn't want to highjack Dave's thread, so I have renamed it and
kept the past emails in it. You have gotten my interest here with your
last response - have you proven that the issue with parked
It should be configured as an unmanaged gateway.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Becker, Jesse
Sent: Thursday, February 23, 2012 6:34 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Patton
software
Subject: Re: [sipx-users] Patton and MOH
I just restarted services and MOH is working over the patton now.
Thanks,
Jes
On Thu, Feb 23, 2012 at 9:35 PM, Todd Hodgen thod...@frontier.com wrote:
It should be configured as an unmanaged gateway.
From: sipx-users-boun
Just went back and reread your description. The information I gave you will
help in configuring the phone, but there is still some wrong behavior on
your incoming calls. The first call should come in on the top button. In
your current configuration, the second call, if working correctly will
.. The config changes you've suggested appear to have addressed
the issue. I don't have the new software polycom setup yet but so far so
good.
On 02/22/2012 05:48 PM, Todd Hodgen wrote:
Just went back and reread your description. The information I gave you will
help in configuring the phone
: Sunday, February 19, 2012 1:59 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] All phone registrations expiring
On Sat, Feb 11, 2012 at 6:53 PM, Todd Hodgen thod...@frontier.com wrote:
There was a note from one of the Dev's in the last 60 days or so that
said that 4.6
I would think that there are many eZuce resellers that are jumping on the
opportunity to show the difference between the open source solution and the
eZuce solution after these questions. It's a great sales opportunity for
eZuce resellers, to show how they differentiate that product from other
call manager for the past 3 years without success.
The guy who runs the IT shop is certified in Cisco, so that's what he wants
to use.
Just my 2 cents.
I truly appreciate your insight, Jes.
Jes
On Thu, Feb 16, 2012 at 9:58 PM, Todd Hodgen thod...@frontier.com wrote:
Tim
With some meals, nothing can replace a good whine.
Especially with a generic bottle with no label of what ingredients are in
it. Leaves you wondering what you just drank that left that terrible
aftertaste.
From: sipx-users-boun...@list.sipfoundry.org
Tim, This type of email is disappointing for me. They highlight negative
experiences that are experienced by a user, and do nothing to help this
project. I'm sure you are being open and honest, but all that it shows is
that you are struggling with the system, in your deployment, and it gives a
I think Best Western was the other hotel recommended. They have a rate for
CSU is you request it.
I am arriving at 1pm on Sunday. If anyone needs a ride to Ft Collins, let
me know.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Is anyone using sipXecs with Juniper switches? I'm looking for validation
references that use Juniper EX switches and LLDP-MED, QOS, etc.
Thanks for any information with regards to this manufacturer you might have.
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I don't want to be contradictory here, but I have many small sites running
on 2gb memory, and 0 swap used in 4-8 months of running that way. These
sites have 5-40 phones on them and they are 100% reliable for me. I'm sure
there is a tipping point to where 2 gb is not enough, but I've not seen it
I'd look at the Firewall as Tony has suggested. Also, ask the ITSP if they
handle Invite without SDP - that was an issue win one large ITSP. They have
created a workaround for it.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Mark
Try a softphone and see if it works correctly, that would at least allow you
to understand what direction the issue is coming from. Do calls into the
auto attendant get answered okay, and transferred?
If the softphone doesn't work, check with your ITSP and see if they accept
Invites without
Change the order of the lines on the phone and see if the problem flips
lines.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Claas Hilbrecht
Sent: Tuesday, February 07, 2012 1:33 AM
To: Discussion list for
Mark,
You will want to provide the name of the ITSP, the model of phones you are
using, and the version of firmware they are running on. Brand of router can
be helpful as well.
Do a search on the wiki at wiki.sipfoundry.org and look up the page on the
sipviewer. It will have instructions
This is precisely why I don't use VOIP.ms as a primary route with customers.
I experience the same thing out of their Seattle server, and I used to
experience the same out of Dallas and Los Angeles. In my opinion, they are
not ready to be a recommended service for customers that use them as
There is a program, Tracebuster, that will show you if you are receiving
sipvicious attacks. For $99, I believe it's a great investment. Simply
monitor traffic from the router, it will show sipvicious attacks, and is
also great for measuring Jitter on a network having issues.
-Original
.
¿Is Fail2ban useful in this situation??
Saludos/Regards
--
Ing. Gerardo Barajas Puente
Ingeniería | www.neocenter.com
T:+52 (55) 8590-9000 x 7003
On Sat, Feb 4, 2012 at 9:33 PM, Todd Hodgen thod...@frontier.com wrote:
There is a program, Tracebuster, that will show you if you are
receiving
I've stood next to you, you are the big brotha'
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Thursday, February 02, 2012 12:50 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users]
Picher
Sent: Thursday, February 02, 2012 1:20 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Sipxconfig-report
hey!
you're no jack flash either :-)
On Thu, Feb 2, 2012 at 3:52 PM, Todd Hodgen thod...@frontier.com wrote:
I've stood next to you, you
Look for a Cisco founder under the tftproot. May not have all of the facts
straight her, but it is similar to this. The new Cisco firmware makes some
changes, that make it necessary to put their files in a directory named for
the model of phone. I believe a fix for this was created and put
One very limited option you might consider is with the dial strings in the
devices. For example, if you limit strings from being dials, something like
|2xxx|[4-9]xxx| should disallow the dialing of 3xxx numbers.
Not that this would be pretty, but I've used similar to control access to a
range
Nathaniel, take a look at the [mac].cfg file for that phone under
var/log/sipxpbx. I think that is a new template within sipXecs. However,
it is of the old style of configuration for Polycom, where you have to
explicitly define all of the variables, versus defaults in the newer
configurations
for users of sipXecs software
Subject: Re: [sipx-users] Monitored Speed dial and directed call pickup on
VVX 1500
What is the setting for the featureDirected call pickup
on the vvx as compared to the other polycoms that behave differently?
On Wed, Jan 25, 2012 at 5:58 PM, Todd Hodgen thod
=
-
It is not a huge issue - it just seems odd that the VVX behaves differently
than the soundpoint.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: Wednesday
Tony makes a good point. Make sure your sub-accounts register to different
switches. IE subaccount one to Chicago, subaccount two to Montreal, etc.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Try creating a Gateway configuration with it in the sipXecs system. Save
the configuration, and then reopen it and click on the INI file that it
creates to save to desktop. It is on the right hand side of the screen.
Log into the Mediant 1000, copy the INI file to it and reboot the device.
If
/18/2012 11:21 AM, Todd Hodgen wrote:
How big of an installation are you talking about in number of users?
Broadvox has the ability to do reroute scenarios automagically in the
event of a failure. They calls could be routed to an analog trunk or two,
which
could terminate in an analog gateway
In the hunt group, it will be forwarded to the voicemail of the last
extension. However, you can create a phantom extension, that forwards all
calls to an express route to any voicemail you would like.
The scenario you have on your current PBX is very workable. Forward
incoming calls to a
, 2012 3:08 AM, Todd Hodgen thod...@frontier.com wrote:
In the hunt group, it will be forwarded to the voicemail of the last
extension. However, you can create a phantom extension, that forwards all
calls to an express route to any voicemail you would like.
The scenario you have on your current PBX
IF you look at the architecture, you have several different applications
running in that server - sipxbridge, sipxproxie, sipx media services, etc.
The needs to route the call through it's proxy, and then it needs to route
it through its sip bridge to be able to get to your ITSP. These various
Here is some new news on the Broadvox Fusion Platform. sipxecs will work on
the Broadvox Fusion platform without issues if you specify the Mitel
configuration. I have a customer working with that configuration for about
3 months now without issue. Do not expect to pass Fax on that
is appreciated.
Regards,
Todd Hodgen
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I think this might work for you - Forward the call to an auto attendant that
plays the message, and then have it timeout with no response to where you
want the call forwarded. Under Auto Attendant Options, change the number of
replays to 0, and shorten the DTMF timeout to a shorter period, then
There's a forum?
/under rock
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Friday, January 06, 2012 2:13 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Sip-Trunk
One nice thing is that there is typically a ton of voicemail space on the
system, since it uses all of the excess disk space. In many systems, where
you are having to by storage size, and limit the space that people have, you
have to manage that space very closely.
Even a simple move the
Scott, I don't think I've seen what Firmware you are running on the Polycom
phones discussed in this thread.I know there are several that have
issues. Personally, I only deploy with 3.1.3c and 3.2.4b at this time. Is
it possible you are using a version that has issues?
-Original
I'm trying to have an application pull up a sipxconfig screen without having
to give username and password by placing it directly into the URL. Or,
accomplishing the same via a rest command. Anyone had luck with anything
similar and have the correct syntax for it?
18:55:11 -0800, Todd Hodgen wrote:
Yes, but what is a virtual PRI? Since PRI is an ISDN standard, what is
the
non-standard derivative that comes out of a Virtual PRI? What is it
exactly?
Is it maybe a PRI that is fed out of device that is actually fed via a T1
with SIP trunks
This email is intended as the beginning of a discussion on the sipXecs users
list to benefit people that are looking for information on types of trunks
to connect to sipXecs. Please consider adding to this document with
relevant facts, and I will make it a wiki document for the benefit of the
Tommy, It's been my experience when emails aren't forwarding that there is
an authentication issue with the SNMP gateway that you are pointed to.
There is a good wiki article that discusses it a bit at wiki.sipfoundry.org
Sendmail does seem to work well, and reliably once the setup
Yes, but what is a virtual PRI? Since PRI is an ISDN standard, what is the
non-standard derivative that comes out of a Virtual PRI? What is it
exactly?
Is it maybe a PRI that is fed out of device that is actually fed via a T1
with SIP trunks on it? If it is, its still a PRI, conforming to
I've used Broadvox with customers for over three years. I can't recall an
out of service for any of my customers on their network. They have built a
carrier grade network, 5 9's availability.
I've used VOIP.ms for three years as well - but can't say the same about
their reliability. They
-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
m...@grounded.net
Sent: Thursday, December 29, 2011 7:08 PM
To: sipx-users
Subject: Re: [sipx-users] flowroute VPRI IP authentication
sigh
On Thu, 29 Dec 2011 18:55:11 -0800, Todd Hodgen wrote:
Yes
I don't believe so. How about setting up an extension with shared
credentials, and use the Personal Auto Attendant with that extension in
voicemail. You might get the occasional email to manage, but could set it
up on a phone and forward them to staff to manage as emails.
Otherwise, it would
Of Todd Hodgen
Sent: Sunday, December 18, 2011 8:04 PM
To: 'Discussion list for users of sipXecs software'
Subject: Re: [sipx-users] Receiving CNAM
Definitely can get Name and Number from Broadvox. I prefer to use their
Legacy network, and you want to use a Static IP address with them. If you
You can definitely have multiple ITSP's on sipXecs. And, you can have
multiple instances of the same ITSP if you plan it correctly. One common
strategy I use is to have an ITSP used with sipxes, that has international
calling disabled. A second ITSP is used for international calling, using
an
This can be done. Let me provide an example. Three companies running on
one sipXecs. Each have their own account with Broadvox. Each Broadvox
account needs to be setup to different IP addresses on their network. With
this you have three different sip trunk gateways. Set the gateway to not
My own Voip.ms account seem to be functioning fine.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Gerald
Drouillard
Sent: Monday, December 19, 2011 9:13 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Anyone suddenly
Definitely can get Name and Number from Broadvox. I prefer to use their
Legacy network, and you want to use a Static IP address with them. If you
get that, you will be fine.
Their new Fusion network can work as well, but they will have to apply the
Mitel configuration on it. I've not
Just checked, it is working for me.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Melcon Moraes
Sent: Monday, December 12, 2011 9:46 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] sipfoundry servers
I've got a spa504G I can test with if you like.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Monday, December 12, 2011 3:33 PM
To: Discussion list for users of sipXecs software
Subject:
...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Monday, December 12, 2011 3:49 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] CiscoSpa plugin testers needed
On Tue, Dec 13, 2011 at 1:44 AM, Todd Hodgen thod...@frontier.com wrote:
I've got a spa504G I can test with if you
If this is a receptionist, the VOP has the ability to build a directory of
DID numbers, and the person who owns them. When the call comes in, it
displays the callerID, number dialed, and the directory lookup for the
greeting.
From: sipx-users-boun...@list.sipfoundry.org
PM, Todd Hodgen thod...@frontier.com wrote:
If this is a receptionist, the VOP has the ability to build a directory of
DID numbers, and the person who owns them. When the call comes in, it
displays the callerID, number dialed, and the directory lookup for the
greeting.
From: sipx-users-boun
Didn't Josh find an issue with RLS and have a fix that reduced the amount of
traffic at one point? Maybe I have my facts wrong, or maybe it is already
incorporated in Software. Josh?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
How about getting rid of the side cars and having them us an xmpp client for
presence - I believe the updates go to the server, but could be wrong.
Might look at it to see if there is difference.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On
Looking for anyone that is currently using the Cisco 7940 phones with
sipxecs, and information with regards to issues they might be having,
firmware they are using, sipXecs version, etc.
Thanks in advance for any input you might have.
Regards,
Todd Hodgen
If you are simply trying to change the ringtone for the extension, use the
web interface. It overrides what is in the downloaded profiles. You can
provide a link to a wav file for the ringtone.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
Example being used by a customer today -sipXecs -- Ethernet --
Audiocodes M1000 -- PRI --- Nortel CS1k
Works great.
I believe I've seen others describe sipXecs --- Ethernet
Patton 4960 PRI NEC PBX.
-Original Message-
From:
The ITSP's handle that with Concurrent Call Sessions. You can receive
multiple calls to the same number.How the PBX responds to that is
another thing.
Let's not confuse line hunting with hunt groups.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
Many firms today deliver a T1 that is sending SIP trunks, and convert that
to PRI with an Adtran or some other IAD device. You receive PRI. The
backbone is based on SIP. Airespring is a good example of that. Locally,
we have Integra.
Yes, in the old days, as Picher described, the Local
You can use an audiocodes gateway or a Patton Gateway for that
functionality.
SIP to the Audiocodes/Patton then PRI to the Legacy PBX. It's done all the
time, and provides a migration strategy.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
, November 15, 2011 12:43 AM
To: Discussion list for users of sipXecs software
Cc: Todd Hodgen
Subject: Re: [sipx-users] Transfer call to another device?
The proxy can definitely know which users are registered. it can access the
reg db directly. what use case in particular are you referring
?
Thanks,
Mark W. Wood
office: (760)202-0224 X2010
Description: New Image.BMP
www.redphonetech.com
From: Todd Hodgen [mailto:thod...@frontier.com]
Sent: Monday, November 14, 2011 10:40 PM
To: 'Discussion list for users of sipXecs software'
Subject: Re: [sipx-users] IVR
Is this happening on the dial by name directory option? I've found that if
you dial more than three characters, sometimes you get a menu of users and
end up pressing one of the options for one of the users in that list. TO
resolve this, I have updated the recording to say please enter the first
I can see where it might be useful to have something that shows if they are
registered to the proxy..
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Joegen Baclor
Sent: Monday, November 14, 2011
NAT only the sipXecs server and not the
phones. This will allow the traffic to/from the server to come and go on
the necessary ports but then phones can use whatever ports they might need.
Mike
On Fri, Nov 11, 2011 at 5:06 PM, Todd Hodgen thod...@frontier.com wrote:
I have an account
I believe VOIP.ms is in the process of updating their switches. I was told
several months ago that Seattle and one other were the two newest. Makes me
wonder if only the old ones work correctly, and their new platform does not?
From: sipx-users-boun...@list.sipfoundry.org
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