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 Beh
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 Ton
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
[mailto:sipx-users-boun...@lis
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: sip
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
[mailto:sipx-users-boun...@lis
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 Excha
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
[mailto:sipx-users-
leshooting?
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:
esday, 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 referri
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 5
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
T
is static 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 wrote:
I have an account that currently ha
users] Multiple ITSP accounts
It sounds like a firewall issue perhaps.
On Fri, Nov 11, 2011 at 5:21 PM, Todd Hodgen wrote:
Great, thanks for validating this.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Frid
I have 6 itsp's running on 4.4in my lab.
On Nov 11, 2011 5:12 PM, "Todd Hodgen" wrote:
I have an account that currently has one ITSP running under sipxbridge. I'm
adding a second one, completely different company, IP, etc. Unfortunately,
when I added it, the current ITSP
I have an account that currently has one ITSP running under sipxbridge. I'm
adding a second one, completely different company, IP, etc. Unfortunately,
when I added it, the current ITSP failed, and neither ITSP shows up as
trying to register under the diagnostics tab. In fact, under that tab,
the
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
[mailto:sipx-users-b
It will not be manageable out of the box. It requires firmware 4.0 and
above. You will have to manually configure it, or so it seems.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Mark Wood
Sent: Wednesday, N
. :)
-
MM
On Fri, Oct 28, 2011 at 21:37, Todd Hodgen wrote:
Are you going to keep us in suspense?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, October 28, 2011 4:22 PM
To: Discussion list for users of sipXecs
It's probably a simple wav file?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kumaran T
Sent: Wednesday, November 02, 2011 10:50 PM
To: Discussion list for users of sipXecs software
Subject: [sipx-users] Author
ustom+Notificati
ons
You can create it however...
On Sat, Oct 29, 2011 at 7:47 PM, Todd Hodgen wrote:
Anyone know where the EmailFormat.properties file went to in 4.4? Has it
been replaced with a different file name, or moved? Not finding it for some
r
Anyone know where the EmailFormat.properties file went to in 4.4? Has it
been replaced with a different file name, or moved? Not finding it for some
reason.
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If you have a Polycom phone there is a way around that though...
On Fri, Oct 28, 2011 at 6:57 PM, Todd Hodgen wrote:
I discovered that as soon as you install Lync with a pc, the ability to use
the TAPI interface is disabled. Real bummer for customers that have the
SIPTAPI for click-to-call
I discovered that as soon as you install Lync with a pc, the ability to use
the TAPI interface is disabled. Real bummer for customers that have the
SIPTAPI for click-to-call.
There is a Knowledgebase article from Microsoft for this particular case of
the TAPI being disabled. The KB ar
You could build some redundancy into your network via a second connection to
your main location. How about low cost DSL circuits, with VPN between the
different routers. When you have a T-1 failure, you would have some
re-route scenarios to limp by.
I suspect having one email platform is pro
Do you have *78 defined in your dial string for the phones? Does it
have a T behind it? I'd play around in that area and see if it speeds
things up for you.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathan
: Friday, October 14, 2011 12:47 AM
To: Todd Hodgen
Cc: 'Discussion list for users of sipXecs software'
Subject: Re: [sipx-users] Outside caller without proper caller-id
Yes,That to make outside calls.My concern is about displaying the caller -id
for outside caller.SipXecs will discard t
You should be able to create a custom dial plan to insert the 00 to make
that call.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kumaran T
Sent: Friday, October 14, 2011 12:00 AM
To: Discussion list for users o
Here is a note from Polycom. It's not a free solution, but it is available
from what they are saying. Don't know about the implementation or
requirements, or compatibility with sipXecs.
"Polycom RealPresence Mobile is the first enterprise software solution for
tablets that lets mobile users e
support it), plus its not
shipping yet. do they have another solution?
On Wed, Oct 5, 2011 at 1:11 PM, Todd Hodgen wrote:
Polycom has a solution.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Wednesday, October 05,
Polycom has a solution.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Wednesday, October 05, 2011 10:07 AM
To: m...@grounded.net; Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Does Bria on
I've seen very similar results when the ITSP cannot accept SDP without
options as well. Check with your ITSP.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Sunday, October 02, 2011 4:29 PM
T
:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, September 29, 2011 10:51 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Centralized Faxing
On Thu, Sep 29, 2011 at 1:46 PM, Todd Hodgen wrote:
Or build a filter into your
Or build a filter into your email client to move the received attachment to
a known folder.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Steve Beaudry
Sent: Thursday, September 29, 2011 10:38 AM
To: Discussion list for users of sipXec
If I'm not mistaken, Option 66 in DHCP tells the phones where the
provisioning files are located. Is it not the case that the Snom phones
just need to put their provisioning files in the same directory where
Polycom's are located, and Snom phones should find them there with the
Option 66 informat
Odd that turning off Caller ID caused this to work. Did you have traces
both ways to compare to see what changed in the call flow? It would be good
to identify where the actual issue is and have it identified for resolution.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun
roadvox does seem to come up on many of the lists I have found:
http://www.siptrunk.org/interopvendors.php
I honestly had no clue that there weren't some high end providers that were
definitely tried and true with Sipx.
On 9/20/2011 1:11 PM, Todd Hodgen wrote:
I would agree with Mike'
I would agree with Mike's assessment here. I've used VOIP.ms for almost
three years now. They are okay, and certainly have improved over the years.
They are much more reliable than they were several years ago. But, I don't
sell them as a sip trunk solution to my customers.
Having said that,
issue.
At the same time, FIOS is two different products. A home user does not get a
50MB upload seed and a static IP like some of our sites do.
At the same time, the FIOS residential modem is a piece of... I mean PITA...
On Mon, Sep 19, 2011 at 2:31 PM, Todd Hodgen wrote:
> Ive had two c
Telecom
except scs is not certified yet either.
On Mon, Sep 19, 2011 at 2:26 PM, Todd Hodgen wrote:
> SO, with TW, nothing would keep you from saying it was an SCS 4.0 install.
> Its not like they are going to ask to look under the hood, they just
> want to understand the pro
I've had two customers using Verizon FIOS, and both had call quality issues.
I have it at my home as well, and experience the same. For whatever
reason, their network is not optimized well for carrying VOIP in my opinion.
I'd love to hear that others have had a different experience.
From: sip
SO, with TW, nothing would keep you from saying it was an SCS 4.0 install.
It's not like they are going to ask to look under the hood, they just want
to understand the provisioning requirements on their end, and that
documentation is in your hands for connecting it up.
From: sipx-users-boun...@
The branch feature works well for this as Mike has suggested. Set up a
branch, and change the location from All to that branch you have created
with the Gateway.
Some ITSP's will have multiple IP addresses to their same switch. Broadvox
is an example of that with their legacy platform. You c
sipx-users] Trouble with setting up SIP trunking
Fyi, dial plans process all calls after user extensions/aliases... not just
outbound calls.
On Sep 16, 2011 12:42 PM, "Todd Hodgen" wrote:
> Tony brings up a good point here, that there really isn't a great document
> describing
s
in the wiki.
On Sep 16, 2011 12:42 PM, "Todd Hodgen" wrote:
> Tony brings up a good point here, that there really isn't a great document
> describing in detail when you look at the overall architecture of sipxecs.
> The dial plans in sipXecs are only used for Outgoing calls,
Mike, we seem to be running in circles in this discussion that doesn't end.
Just to be clear - If you set up an IPsec tunnel between these two
locations, and attached Polycom phones to the remote network, they will
register onto your sipXecs server as if they were sitting physically in the
same bui
Tony brings up a good point here, that there really isn't a great document
describing in detail when you look at the overall architecture of sipxecs.
The dial plans in sipXecs are only used for Outgoing calls, and have no
effect on incoming calls. And, just to re-iterate what I've seen answered
he
Setup an IPSEC tunnel between the two locations - you should have all of the
sipXecs features. Except 911 calls dispatching to the right location. Put
in a local FXO gateway for that.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoun
Mike, You've entertained that discussion on several occasions during the
past. From my perspective you are so focused on a single box solution for
whatever reason, you miss the important part of it - creating a reliable
platform for voice. Over the years, it's no secret that those that run
data
ing a PC... i guess it's just me. i can
replace it in a snap, i know where to get power supplies, etc.
On Tue, Sep 13, 2011 at 1:22 PM, Todd Hodgen wrote:
$214 buys a PFSense appliance loaded with three ports from Netgate. Works
great!
From: sipx-users-boun...@list.sipfoundry.org
[ma
Oops, I guess my response was from the office of redundancy.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Tuesday, September 13, 2011 9:56 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-use
$214 buys a PFSense appliance loaded with three ports from Netgate. Works
great!
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Tuesday, September 13, 2011 9:44 AM
To: m...@grounded.net; Discussion list for users o
You should be able to adjust the ports within sipXecs to accommodate that
provider. Although, there has been a lot of chatter on this list with
regards to people not being able to set things up with Flowroute. You
might want to go back into the archives to see how they worked around the
issue.
Anyone had experience using Simple Signal as an ISP with sipXecs? Comments
good or bad?
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List Archive: http://list.sipfoundry.org/archive/sipx-users/
How about putting a Gateway at the far end site, via a VPN. The phones
would register as part of the main system, and calls would be routed however
you configure them in that gateway - local trunks, or trunks from the PBX at
the far location.
QOS considerations need to be engineered into the d
red
and have no clue who built it or its current state.
Mike
On Fri, Sep 9, 2011 at 7:25 PM, Todd Hodgen wrote:
I know this is a stretch, but does anyone on the list use Mitel phones with
sipXecs? I understand their SIP is not very compliant, but never had
firsthand experience with the
vices rebooting and attaching to wrong vlan
even after the vlan was manually set and the config file stated it, then i
chucked 'em. at least i got a workout.
On Fri, Sep 9, 2011 at 2:17 PM, Todd Hodgen wrote:
With the SPA942, the phones work great. The only issues I'm seeing right
now
I know this is a stretch, but does anyone on the list use Mitel phones with
sipXecs? I understand their SIP is not very compliant, but never had
firsthand experience with them. Anyone have real world experience with
them? I have a potential customer with 16 of them, we are trying to
determine if
With the SPA942, the phones work great. The only issues I'm seeing right
now is Shared Line Appearance not work, WIKI implies it's not supported so
that is okay, and the clock doesn't display correct time - seem to display
GMT, rather than local time. Need to dig into that still.
MOH, transfer,
Shared line appearance was only tested with Polycom and Snom from what I
recall on the wiki.
I've been trying to get it to work correctly with SPA942 without success.
Admittedly, I haven't put a lot of effort into it yet.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
I have two installations running on 4.4 with patches that are using the
successfully using the Intercom feature. One with Polycom, one with Linksys
SP942 phones.
No customization has been done to software other than a dropped in java file
for sipxconfig.
From: sipx-users-boun...@list.sipfo
I have just hired a developer to create the ability to convert TIFF files to
PDF within sipXecs. I expect it to be ready in the next 14 to 21 days. In
its current development, it will be an all or nothing - the system all gets
TIFF, or the system all gets PDF. It will be donated to the community
Counterpath seems to be a company struggling in many ways. Their reseller
program was sunk over a year ago. So, resellers such as myself has no
motivation to promote their product. Of course the support issues are now
famous.
It may well be time to start collaborating towards integration wi
Bad day?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, August 26, 2011 1:21 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Problems remain after linear upgrade
no it wont,
What routing is sending these internal calls to the one phone? Something
in call routing is putting the calls to his extension. I would identify a
phone number that he is getting calls for, look to see where that alias is
assigned in the system, and follow the routing for it. You will find his
sends
the INVITE back to the gateway.
Kyle
On Tue, Aug 23, 2011 at 10:15 AM, Todd Hodgen wrote:
> Kyle, Just to confirm your setup is workable with sipxecs, I've set
> that scenario up with an ITSP and sipxbridge, and it did work correctly.
>
> By chance is your Audiocodes set
Kyle, Just to confirm your setup is workable with sipxecs, I've set that
scenario up with an ITSP and sipxbridge, and it did work correctly.
By chance is your Audiocodes setup with the IP address of sipXecs instead of
the FQDN? I recently had a similar issue with an Eqygi gateway that was
fixed b
, since I'm not logged into the SipXecs right now, how do we run a
> diagnostic configuration test.
>
>
>
> Brian Buckles
>
> IT Manager
>
> IT Network Consultants
>
> (859)963-1911
>
> 877-888-ITNC (4862)
>
> brian.buck...@itnetworkco
er
>
> IT Network Consultants
>
> (859)963-1911
>
> 877-888-ITNC (4862)
>
> brian.buck...@itnetworkconsultants.com
>
>
>
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
>
> Sent: Sat
Look up sipviewer on the wicki - it will be your friend - downloand and
install on a computer.
Make sure you have Winscp on your computer as well.
Make sure you have Putty.
Go to /var/log/sipxpbx on the server and do a logrotate. Instructions on
the wiki.
Try a call, and then run merge-logs
ng ones. I would imagine if you
issued them, it could cripple sipx since that might interfere with its own.
Does the ISP you use not have a magic host you could use instead? The ISP
blocks sending email?
On Fri, Aug 19, 2011 at 1:32 AM, Todd Hodgen wrote:
I believe a few on the list are u
.
Does the ISP you use not have a magic host you could use instead? The ISP
blocks sending email?
On Fri, Aug 19, 2011 at 1:32 AM, Todd Hodgen wrote:
I believe a few on the list are using Gmail or Google Apps. Any success
setting up Gmail for SMTP gateway to sipXecs and any pear
I believe a few on the list are using Gmail or Google Apps. Any success
setting up Gmail for SMTP gateway to sipXecs and any pearls of wisdom for
setting it up?
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This is a large ACD. To use a new, or newer product for that type of
environment seems like suicide to me. Regardless of how good it is, if you
are talking about a 24 hr shift, with just 5 minute calls per agent, you are
talking about an average of 416 calls at one time. Personally, I'd be
look
Sorry, I missed the ACD part of this question, disregard this response.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: Wednesday, August 03, 2011 8:42 AM
To: 'Discussion list for users of sipXecs software'
S
When you load multiple files, it plays them all like a jukebox.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Max DiOrio
Sent: Wednesday, August 03, 2011 8:07 AM
To: Discussion list for users of sipXecs software
(sipx-users@list.sip
Unless I'm missing something, that is a simple Auto Attendant with an alias.
People dialing that alias would hit that particular Auto Attendant. The
message you put on the Auto attendant, and the options you provide control
how it is used. "Hello, you've reached Mike's Youth Hostile, Dial the
ext
] Spectralink 8400
3.3 is not tested with sipx yet at all...
On Mon, Aug 1, 2011 at 8:02 PM, Todd Hodgen wrote:
I was told they would be on 3.3+
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Monday, August 01
n the Soundpoint sip
stack?
On 7/21/2011 9:35 PM, Todd Hodgen wrote:
I am supposed to be receiving some 8040's for test and evaluation. I'll add
something to the wiki when I get it completed. I have not received a
timeframe yet though.
From: sipx-users-boun...@list.sipfound
You can build a 2 or 3 port VLAN, and use it to run your internet line into,
and then another port out to your ingate. Port mirror either port to get
the traffic ahead of the ingate, or use the third plug for your wire shark
device.
-Original Message-
From: sipx-users-boun...@list.sipfoun
To take a late night stab at this, you will need to have the numbers defined
in the system somewhere, either as an alias on an extension, or as an alias
in one of your auto attendants.
If you don't define them as an alias, and your sipxbridge is not set to
default calls to the auto attendant, y
i had deployed with
VLans and was able to put a firewall between data and phone vlans and the
problem went away...
On Mon, Jul 25, 2011 at 11:15 AM, Todd Hodgen wrote:
I've had several SPA942's working in my lab for a few weeks, I did not see
them reboot on their own at all during that time.
I've had several SPA942's working in my lab for a few weeks, I did not see
them reboot on their own at all during that time. I actually used one of
them on my production system running 4.2.1 as my own phone for about a week
and did not experience this particular issue.
These are now at the custom
Is there a particular reason you are moving to 6.0? 5.6 should work fine
with sipXecs.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rhon
Sent: Sunday, July 24, 2011 8:25 PM
To: Discussion list for users of sipXecs software
Subject
That is indicative of an issue in and of itself if they can't be restarted
by the Config server.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Max DiOrio
Sent: Friday, July 22, 2011 9:47 AM
To: Discussion list f
Broadvox as well, but only on their Legacy Platform.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Friday, July 22, 2011 8:51 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] A good ITSP
Are you referencing the BCM 50 or the CS1000 from Nortel/Avaya? These are
two different beasts, with completely different requirements. Nortel
created instructions for configuring a BCM 50/450 with the SCS (Nortel
Commercial version of sipXecs).
Martin can probably give details with regards t
I am supposed to be receiving some 8040's for test and evaluation. I'll add
something to the wiki when I get it completed. I have not received a
timeframe yet though.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Paul Curtis
Sent: Th
Your registration will go out over Port 5060, and it requests it be returned
on 5080. If your request comes back on 5060, it will go to the proxy, and
not to the sipxbridge. Ensure it returns back to you on 5080.
On the Wiki, get familiar with sipviewer, and the use of the merge-logs, and
si
I can confirm that I have 12 users in a huntgroup that all receive a call
when someone rings a doorphone. It has worked flawlessly for over a year.
That doorphone is used at least multiple times each day. I would say Mike's
answer is spot on.
From: sipx-users-boun...@list.sipfoundry.org
[mail
Sounds like a great release. Thanks for your hard work!
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jean-Hugues
Royer
Sent: Thursday, July 14, 2011 6:11 AM
To: Discussion list for users of sipXecs software
Subject: [sipx-users] Voic
I am working with some SPA942 phones now. MOH works fine in a traditional
and expected way - place a call on hold. ON the wiki there are some notes
on use of the SPA942.
After hearing a lot of negative on these phones, I'm quite surprised at how
well they actually work with sipXecs.
From:
Here is an article where someone implemented it in Asterisk. Same principal
should apply with the Polycom config in sipexec if you want to play around
with the file.
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
Just remember, if you send the profile after you have changed it,
If you use the siptapi application (open source app) within outlook to do
your dialing, there is an option field for User's Extension which is the
extension it dials for your click to call. In that field, put *76+extension
number and it will call you via intercom call, answering automatically, and
Maybe I'm mistaken, but I think there is an error in this response.
" note: if you send REGISTRATION to the itsp, it will come
FROM port 5080, and they should RESPON on port 5060.
if you use ip based authentication, and not registration,
set everything on sipx back to normal (might need to delet
Mike - I think you point out an important feature of sipXecs that many
people don't find - the line configuration. I suspect many don't know that
you can select each line on a phone and have a separate configuration on it.
I personally didn't realize this for quite a while.
By selecting the l
If you look at a packet capture, you will see the information is there.
SipXecs forwards the information to the end point I suspect. But the end
point needs to display what you are asking for. I could be wrong, but I
don't think this is a sipXecs issue.
From: sipx-users-boun...@list.sipfound
Why could you not assign an extension to each user - which is their primary
extension number, and have DID and other aliases point to it. Have another
number that the hunt group points to. Line 3 rings, its hunt group. Line
1 or 2 ring, it's a call to your extension. Pretty simple.
From: s
The capability is there, its the end point that needs to handle it.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Anthony George
Sent: Thursday, June 30, 2011 1:45 PM
To: Discussion list for users of sipXecs so
Voice Operator Panel will do exactly this. It will even open up a directory
of users within that called number. Say you have 5 companies. You have a
director of 20 users in each company. When call comes in for Company ,
directory of 20 phones for Company will be shown.
It will allow
301 - 400 of 963 matches
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