gateway to that Dial Rule.
Now, that gateway, when selected, can use those dial plan rules for dialing.
From: Michael Scheidell [mailto:scheid...@secnap.net]
Sent: Monday, January 18, 2010 10:30 AM
To: Todd Hodgen
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] how do I diagnose
You can disable access to all gateways on an individual phone, and then
enable just the one you are working on to test it. The question will be is
it accessing the gateway and getting denied by the gateway, or is it getting
denied by sipXecs because of a dial string issue, lack of permission, etc.
users will have to be in the assigned group to access it.
That is where I would start looking.
-Original Message-
From: Andrew Cotter [mailto:andrew.cot...@somersetcapital.com]
Sent: Sunday, January 17, 2010 7:24 PM
To: 'Todd Hodgen'; 'Josh Patten'
Cc: sipx-users@list.s
Nice catch Josh, thanks.
-Original Message-
From: Josh Patten [mailto:jpat...@co.brazos.tx.us]
Sent: Sunday, January 17, 2010 2:21 PM
To: Todd Hodgen
Cc: 'Andrew Cotter'; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] AudioCodes MP-118 and SipX
You mentioned you a
Do your configuration tests all pass (Diagnostics/Configuration tests)?
Does a Preflight test pass as well?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of John Buswell
Sent: Sunday, January 17, 2010 10:56 AM
To: Tony Graziano
Cc: sipx-u
You haven't indicated how you have programmed the MP-118, so not clear if
you manually configured it, or used a configuration from sipXecs. And, is
this a purely FXO product and not mixed fxo/fxs?
My recommendation is to start with Sipxecs and configure the MP-118 with
their template. Once this
p you with figuring out how you transition from one system to the
other.
Haven't worked with Skype, so I don't know their specifics.
Hope this helps.
From: Michael Scheidell [mailto:scheid...@secnap.net]
Sent: Saturday, January 16, 2010 11:52 AM
To: Todd Hodgen; sipx-users
p.net]
Sent: Saturday, January 16, 2010 11:30 AM
To: Todd Hodgen
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Trying to regester an external line cisco 7960
My first shot at sipx, so, let me make sure I do it right.
I can't set permission on a trunk(gateway), but can create
: Todd Hodgen
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Trying to regester an external line cisco 7960
My first shot at sipx, so, let me make sure I do it right.
I can't set permission on a trunk(gateway), but can create a private group
and just put that user in the group,
How about making that external line just another trunk, and then give
permission to use that trunk just to that phone by putting it in its own
group? It might be a simple workaround.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Micha
Tony, I think that Geoff has given you a good solution. Go to Auto
Attendant, define a new auto attendant. Go to Dial plan, build a new dial
plan, using the auto attendant, assign it an extension number. Then, you
can forward no answer to that extension number, which is an auto attendant.
Should
Look at Features / Auto Attendants / Options. Transfer on Failures.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
mkitchin.pub...@gmail.com
Sent: Thursday, January 14, 2010 8:42 AM
To: sipx-users@list.sipfoundr
Actually, if you look further, D is an invalid option.
FramingMethod_x
(Framing Method)
Same as "FramingMethod" for a specific Trunk ID (x = 0 to 7)
Their documentation is a bid confusing, I guess you have to read more of it
in detail.
-Original Message-
From: sipx-users-boun...@list.sip
According to the Audio Codes documention, 0 & D both specify Extended
Superframe with CRC6. Are you sure that was your problem?
FramingMethod
(Trunk Settings>Framing
Method)
For T1
0 or D = Extended super frame with CRC6 (default)
1 or B = Super frame D4, F12 (12-Frame multiframe)
A = F4 (4-Fr
In the configuration of the phones, did you set them to domain name, or
Fully Qualified Domain Name. It needs to be Domain name, it will get the
FQDN from our srv records in DNS. Personally, I'd look there.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundr
There is a ton of development going on with sipXecs today. The developers
list is very active with work that is being done. The roadmap at sipXecs
outlines what is being done as well.
There was an announcement on this list just a few weeks ago with regards to
Avaya's commitment to the project
Doing that with Polycom 450 phones. No issues. Two lines per phone.
V3.1.3c split from pcom.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Picher, Michael
Sent: Friday, January 08, 2010 1:31 PM
To: sipx-users@list.sipfoundry.org
Subj
Are you not able to log into the voicemail using their extension number and
password via a telephone to change the greeting? If you don't know their
password, you can change it in their profile and then access the voicemail
box from any telephone on the system, without having to get down into the
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, January 07, 2010 11:10 AM
To: Eric Varsanyi
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] [sipX-dev] Handling firmware for Polycoms in a
mixed
Why not have the G722 as first option, G729 as second option, and then G711
as third option. This should put internal G722 to G722 compliant phones
working on that CODEC. Then, when they leave the system to systems that
don't support G722, they will go to G729 next. The devices will do their
own
Nathan,
The best thing to do right now is to get a complete trace of these failed
calls. Clear your logs, regenerate the failed calls and grab a merged Xml
file. With that, I'm sure someone on this list will be able to point you in
the right direction fairly quickly. My guess is that you do
Voip.ms does work with version 4.0.x as well as version 4.1.x. I've used it
with both on several servers, as that is what I test with. I find that I
get better results with them if I use their IP address rather than their
FQDN. Under Devices/Gateway, ITSP Account, I have all of the defaults
mark
ntial problems.
From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
Sent: Saturday, January 02, 2010 5:50 PM
To: Todd Hodgen; Chris Rawlings; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Fwd: Inbound Call not hanging up
That's a method I use for routing to a live attendant
Why do you have the Alias answered by a Phantom Ext, and then forward to an
Autoattendant? You can place that Alias number in the Auto-Attendant so
that it goes directly to the Auto-Attendant and accomplish the same thing.
Doubt if this is related to your issue, but seems like an extra unnecessary
Yes, you can import the user accounts. Create one user account, assign them a
phone and then export the list. That will give you the format. There are also
instructions on the right side of the screen. Go to System, Import/Export.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sip
Octothorpe? I've not heard it called that Scott. I love it. Me thinks
you've got the Telecom Dictionary in your Library.
-Original Message-
From: sipx-dev-boun...@list.sipfoundry.org
[mailto:sipx-dev-boun...@list.sipfoundry.org] On Behalf Of Scott Lawrence
Sent: Thursday, December 3
It could be many things, and speculating for you won't fix the issue. Last
week I had that error because of a MOH issue.
Hear it is in a nutshell - Client Error (4xx): The request contains bad
syntax or cannot be fulfilled at the server.404 specifically is Not
Found.
You will need to do a tr
There is nothing really helpful here that would allow for any
troubleshooting for your issue.
The fact that calls don't go out or in is an indication of an issue that
will be logged. What is in those logs will point to where the issue is.
When calls are not going out, you will need to make a call
I'd start by looking at that import file to see if there are any other
trailing or leading spaces, etc. I'd also look at the format of that number
you are trying to dial, try taking out the parens, the hyphens, etc. to get
to just digits, and then ensure that those digits are in a dial plan
somewh
e ever had this issue on
any other version before.
On Thu, Dec 24, 2009 at 12:57 PM, Todd Hodgen wrote:
Chris,
Try to configure that AA to call a phantom extension. Have that Phantom
extension ring as a forward to those 4 phones instead of using the hunt
group. This might work for you rathe
Chris,
Try to configure that AA to call a phantom extension. Have that Phantom
extension ring as a forward to those 4 phones instead of using the hunt
group. This might work for you rather than the hunt group.
Can you explain the ACD service in more detail. Is that an issue related to
this hun
Luckily, there is an unlimited supply of extension on the PBX. Cant beat
the price either.
From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
Sent: Monday, December 21, 2009 11:44 PM
To: Todd Hodgen; Rene Pankratz
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Is a user
button on the phone to forward the main
extension to the secondary extension with the voicemail box.
From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
Sent: Monday, December 21, 2009 11:23 PM
To: Rene Pankratz; Todd Hodgen
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Is a
at the top, you will need to put in the 4+first
digit of extension, and then assign your dial plan that was assigned under
system/dial plan.
-Original Message-
From: steven warner [mailto:stevenwar...@gmail.com]
Sent: Monday, December 21, 2009 10:50 PM
To: Todd Hodgen
Subject: Re: [
merge-logs
This will create a merged.xml file. Open that file in sipxviewer.
Are you sure the ports are open in your firewall for these calls?
-Original Message-
From: steven warner [mailto:stevenwar...@gmail.com]
Sent: Monday, December 21, 2009 10:26 PM
To: Todd Hodgen
Subject: Re:
it worked fine.
-Original Message-
From: steven warner [mailto:stevenwar...@gmail.com]
Sent: Monday, December 21, 2009 8:51 PM
To: Todd Hodgen
Subject: Re: [sipx-users] Connect two SipXecs systems together.
Hi Todd,
I must be missing something. If Site A calls site B,
how does Site B kn
Start with your DNS. Can you see the the Other side from each site?
When you connect these site to sites up, they are assumed to be trusted, so
no password is required. You couldn't get in if they didn't explicitely let
you in.
Your dial plan seem accurate.
-Original Message-
From: sip
If you make changes to an individual phone, it will overwrite your Group
Configuration. If you configure it via a browser, it will overwrite the
sipXecs individual configuration.
It actually works very well at a group level, and no configurations are
required at an individual level unless you
You can buy boat anchors on Ebay...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of shouldbe q931
Sent: Sunday, December 20, 2009 5:48 PM
To: sipx-users
Subject: Re: [sipx-users] Avaya and sipXecs
On Sat, Dec
I see a problem here worth considering. If sipXecs is designed as a
business telephone system, then the business owner needs to be able to
maintain some control over their business, and how the business uses the
telephone system. By giving the end user access to any and all controls,
you are effe
Voice Operator Panel works nicely. $400. Supports a tethered phone, which
provides for much higher voice quality than running it on a PC. Seems to be
a solid product. http://www.joher.com/voiceoperatorpanel.shtml
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@
Some basic understanding here is in order. When you set up your ITSP bridge,
all of the calls are set to go to the operator. Remove the Operator from the
SBC configuration for SipXbridge.
Next, you need to assign the main number. If you want it to go to the
autoattendant, then configure i
Jordon,
You won't be able to assign that DID directly to a conference extension, as
they don't have an alias configuration available. You can create a phantom
extension, assign the DID to it as an Alias, and then set up a forwarding
rule for that extension to your conference number. Works great
I recall that someone else was having this issue in the past, and they were
able to get Vitality to change the port for them.
You need things to arrive on 5080 if you want the SipXbridge to be involved, to
support thinks like Refer.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:si
Yes, Voice Operator Panel does have several versions of their product. One
is a PC Console, the others are slimmed down products that are licensed for
individual users desks that provides click to call, presence, etc.
I believe those features are part of the XMPP implementation as well, but I
You see this a lot in CDR. When a call is transferred to Voicemail and the
drop off, or when they drop out of the Auto-Attendant you see failed calls.
I'm unclear if it is because the call length was too short, or what, but it
seems to be a normal behavior today with sipXecs.
If someone knows
One very simple way to create your messages for sipXecs and ensure they are
formatted correctly is to leave a recording in a voicemail box, and then
grab it from the user portal as a wav file. Its guaranteed to be in the
correct format, and its very simple to create and move to your recordings.
Jake,
You can make adjustments to fix this in the dial plan under the phone itself
- there is a digit mapping form. You need to make some considerations here.
Most likely you will want to do this under the Phone Group, so that it
applies to all phones of the same type. I'd recommend you find
It is nice that they summarize the calls in progress and the trunks used.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Scott Lawrence
Sent: Tuesday, December 08, 2009 7:10 AM
To: Jordan Turner
Cc: sipx-users@li
: Tuesday, December 08, 2009 3:32 AM
To: 'Todd Hodgen'; 'Sipx-users list'
Subject: RE: [sipx-users] PC Console for sipXecs
Todd,
I didn't find pricing info on their site.
Can you shed some light on how much does it cost?
Thanks in advance,
Nikolay.
_
I've recently had the opportunity to work at length with a PC Console by the
name of Voice Operator Panel. A trial version can be downloaded at
http://www.voiceoperatorpanel.com/trial/VoiceOperatorPanel-setup.exe
This console is very simple and straightforward in how it works. Getting
presenc
LOL!
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Scott Lawrence
Sent: Monday, December 07, 2009 8:23 PM
To: Tony Graziano
Cc: Sipx-users list
Subject: Re: [sipx-users] using regexp in dialing plan, does anyone
n somewhere that has more than a single route, with
custom dial plans on a per phone or group basis, right? With a defined dial
plan for the entire network, right?
-Original Message-
From: Matthew Kitchin (public) [mailto:mkitchin.pub...@gmail.com]
Sent: Monday, December 07, 2009
ecember 07, 2009 7:41 PM
To: Todd Hodgen; 'Sipx-users list'
Subject: Re: [sipx-users] Dialing plan with polycom phones
I wish, but I think I'm going to have to mirror the existing functionality.
I hope that is possible.
--Original Message--
From: Todd Hodgen
To: mkitchin.pub...@
Don't know the politics you are dealing with, but it might make sense to
develop dialing policies within your organization. As an example, our
company now does 10 digit dialing for all calls, except International and
and xx digits for internal.
-Original Message-
From: sipx-users-boun...@
There is a simpler way of doing this. User logs into the system with their
user name and Password from any browser, enters call forwarding
configuration and simply puts in the extension number or other number they
want their calls forwarded to.
No development required. You can have it working
Sorry, just reread this question. I believe you will have to define both
networks on the server as Intranet networks. Your phones will be NAT to the
server, but there isn't anything you will have to do. Remote workers is a
different subject.
-Original Message-
From: sipx-users-boun...@
If the Server is on a Public Address, and the phones are on NAT addresses,
how would they get to server without going through a router to get onto the
same network as the server? That is where your Nat will come in -
traversing the router. Seems the Server would need to know they are NAT, as
it w
I'd try Mike's suggestions. The 450 was available prior to 4.0.4, abut you
could use them using the configuration for the 550 phone.
I'd try the suggestion to configure them as 330 phones as an interim
solution.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx
I'm answering this in a format that can be used as a reference for others
who might have this same question down the road.
To answer this in a basic form for those that are not aware of this
terminology, there are two terms you want to get familiar with.
FXO and FXS.
FXO - Foreign eXchange Offi
Ah, thanks for clarifying.
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Friday, December 04, 2009 6:53 PM
To: Todd Hodgen
Cc: Burden, Mike; Tony Graziano; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Disabling Music on Hold (Was: Re: Hold/Resume
One other thing to consider is the release of sipx you are running. You can
leave that on with 4.0.4, at least that is what I understand from a previous
note from Ranga. Are you running 4.0.2 by change, or something earlier?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun
I believe the calling name will be provided by your telco, as they hint in
their reply. It appears they apply a name on a trunk by trunk basis, which
is in their CNAM database, not in yours.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf
If you make that unused Windoze server another instance of sipXecs, then you
can do sipXecs-sipXecs trunking between these two offices, and begin to lay
out your extended dialing plan for all of these offices.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-use
If Verizon provides the ds3 internet pipe, then you can chose any ITSP for
providing your SIP trunks. Other carriers are more user friendly with
regards to supporting other ports, etc.
And, many have user portals that allow you to control your destiny a bit
better.
-Original Message-
Fro
One Google Search of that exact error, and the second result was this.
http://ocsguy.com/2009/08/
The issue is with the configuration from your carrier for the PRI, and the
fix is within your audio codes.
It is unrelated to sipXecs.
-Original Message-
From: sipx-users-boun...@list.sipfo
Can you call from phone to phone on the sipXecs? Can you call from Phone to
Auto-Attendant with just sipXecs? I would want to confirm that the sipXecs
system is working properly before I tried to interconnect to another system
of any type. Did you run the pre-flight and what were your results?
Can you clarify. This is an open source product. The commercial product
does have other online resources that anyone is free to pay for as an
authorized distributor. However, Anyone in this user group is free to start
a forum for support of this product. It's not a responsibility of the
develo
The "Terminal Adapter" in this case will be a Channel Service Unit or CSU.
They are required on the line, many people don't use them, which should not
be the case. They ensure 1's density on the T-1, so you avoid issues on the
span itself from timing slips, etc. It also provides a loopback point
Yes, you can do that. DID is routed to a particular line when you put it in
as the alias. Assign that line to a button on a phone, and it will ring on
that button, on that phone.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.or
7;s just go to busy so for the heck of it, we
set up a phone
on sipx with the alias of the number above. Even though it's got the wrong
area code, it
rings into the phone when we set one up.
On Tue, 24 Nov 2009 12:27:43 -0800, Todd Hodgen wrote:
> Check your setup on the gateway.
What a shame, the discussion I have seen was on March 31. Problems with 100
phones in a group, and trying to add more.
-Original Message-
From: Josh Patten [mailto:jpat...@co.brazos.tx.us]
Sent: Tuesday, November 24, 2009 3:11 PM
To: Todd Hodgen; sipx-users@list.sipfoundry.org
Subject
There was discussion around this on the list a few months ago, and the
technical details were provided by one of the developers. There is a limit,
I don't recall what it is, but it wasn't as high as you are wanting to go
from what I do recall, but it was higher than 12. I'd check in the old
archi
Check your setup on the gateway. Those DID calls need to be sent to sipXecs
with the DID number as the URI, or it will route the call to the URI, which
is probably your main number. Scott Lawrence pointed this one out last
week. SipXecs routes on the uri only.
-Original Message-
From: s
Cisco is Polycom's largest customers - not such a surprise.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Robert B
Sent: Sunday, November 22, 2009 4:45 AM
To: Matt White
Cc: sipx-users@list.sipfoundry.org
Subjec
Jordan,
Great news. I think you have the "ITSP cheatsheet", if you don't just
google it, you will find it on the sipXecs site. Follow it step by step,
literally. Your ITSP will come up and work for you.
The ITSP is no mystery, and Ranga has written up some nice tips that will
help you get it
Jordan,
The emails you called lectures are not that, but good sage advice, not from
users, but the actual developers of the components you are having trouble
with. If you listen to them, and describe the issue you are having in the
manner they are requesting, you will get the best possible suppor
Not a request. You create it and donate it to the project. The beauty of
open source is you can customize it to do whatever you want it to do, you
just have to donate what you create back to the community. When you ask for
features changes, you are asking someone to donate their time to create i
Are we mixing up the feature codes one would use with a Cell phone or
residential Centrex type service with the Features that are available in a
SIP PBX. Surely, one can't expect them to be the same.
However, if someone is trying to create a commercial offering with sipXecs
that includes those
The closest thing I have seen is in Mike Picher's book - he has a sample
user guide that lays out the default feature codes well. There are default
numbers in the system, which can be changed. Like *76, etc.
Mike did a great job of laying them out in a useable format.
From: sipx-users-b
Maybe a similar scenario can be built into software only? A loopback of calls
from bridge to bridge that acts as a choke?
-Original Message-
From: Boy Aidil Sjam [mailto:aidils...@prawedanet.co.id]
Sent: Monday, November 09, 2009 7:24 PM
To: 'Todd Hodgen'
Cc:
.
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: Monday, November 09, 2009 8:50 AM
To: Todd Hodgen
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] FW: unable to route alias correctly
On Mon, 2009-11-09 at 08:11 -0800, Todd Hodgen wrote:
> Th
grazi...@myitdepartment.net]
Sent: Monday, November 09, 2009 8:24 AM
To: Todd Hodgen
Subject: Re: [sipx-users] FW: unable to route alias correctly
And as I look at the invite seen from the proxy it show as NPANXX (10
numbers, not +1 in front of it).
So the aliases you have entered are the full 10
This might sound a bit hairbrained, but
For new offices, could you place an FXS gateway on site, and then loop the
calls right back to an FXO gateway to the switch. Set up a sip trunk that
rings to a hunt group for two fxs ports, which ring to your FXO ports. This
would limit the numb
Yes, it is empty.
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Monday, November 09, 2009 7:55 AM
To: Todd Hodgen
Subject: Re: [sipx-users] FW: unable to route alias correctly
OK. That's a little clearer now. Are you using sipXbridge? If so, do you
have send all cal
nt: Monday, November 09, 2009 7:52 AM
To: Todd Hodgen
Cc: 'sipXecs users'
Subject: RE: [sipx-users] FW: unable to route alias correctly
On Mon, 2009-11-09 at 07:38 -0800, Todd Hodgen wrote:
> The call was hung up when the console began to ring. That number
> should not ring at the consol
ttendant to try to ensure it does not ring the
console. The console is rang only when calls go directly to 827-8255.
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: Monday, November 09, 2009 4:26 AM
To: Todd Hodgen
Cc: 'sipXecs users'
Subject: R
riginal Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: Monday, November 09, 2009 4:29 AM
To: Todd Hodgen
Cc: 'sipXecs users'
Subject: Re: [sipx-users] FW: unable to route alias correctly
On Sun, 2009-11-08 at 23:03 -0800, Todd Hodgen wrote:
> I'm provi
.
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: Monday, November 09, 2009 4:26 AM
To: Todd Hodgen
Cc: 'sipXecs users'
Subject: Re: [sipx-users] FW: unable to route alias correctly
On Sun, 2009-11-08 at 23:03 -0800, Todd Hodgen wrote:
> I&
I'm providing a link to several files from a system where I am having some
issues with the routing of alias calls.
Setup is 4.0.3, with a listed number of xxx.xxx.8255, which is an alias with
forwarding to two consoles. Calls coming into this main number work as
designed. However, any DID cal
It's covered in detail in their installation manual - using MEGACO
specifications.
www.audiocodes.com/filehandler.ashx?fileid=73916
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Geoff Van Brunt
Sent: Thursday,
Also, there is an option in devices under your phones that might be helpful.
Impossible match handling is defaulted to allow "user to dispatch" When
there is a mismatch in the dialing plan, you can hit send to send it out.
If you change that to give reorder tone, and disable some of the gateways,
Has anyone had experience with older Watchguard firewalls not being able to
forward sip packets? IF so, did you find a resolution short of replacing
it?
Situation with a Watchguard firewall passing SipXbridge Invite to an ITSP,
ITSP returns the Unauth, which never gets forwarded through the ro
Another thought here -
Hanging a laptop with a soft client off of the system might be an easier way
to get the integration going rather than the conference bridge. It's just
another extension on the system.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-use
I have not seen a release of 4.0.3 yet. The current stable release as far
as I know is 4.0.2
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Monday, October 26, 2009 6:25 AM
To: sipx-users@list.
This entire discussion is about routing, and not really related to sipXecs.
Once you get the routing to your server, for ANYTHING, you will be much
further along and can then start to work on getting sipXecs up and running.
William, your traffic will arrive at your router/firewall/DSL (router)
int
Check under System, Internet Calling. Confirm in the Intranet Subnets that
you have the correct subnets in there, with the correct subnet mask. Remove
any that are not used but there by default.
That's what I would check first.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-u
An evaluation of the features is all very nice. However, the features are
worthless unless the call quality is there. Did your evaluation include an
evaluation of call quality, dropped calls, blocking rates, number of trunks
supported, or anything technical that demonstrates the quality of the sy
How about having them enter the full extension number for a conference
bridge from the auto attendant, rather than give them the single digit
prompts. You can assign the conference bridges with three or four digit
extension numbers to them.
-Original Message-
From: sipx-users-boun...@list
I see it now, Mike has a part number for a small UPS for that phone - added
value, added sale!
Great ideas.
BTW, if you run into any Nortel BCM's, my nephew has now done
interoperability between them and sipXecs. I'm working on a document for it
now.
-Original Message-
From: sipx-users-
801 - 900 of 963 matches
Mail list logo
