Queuemetrics works well for this also, and can be installed on a separate
machine/VM.
www.queuemetrics.com
Bruce
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz
Sent: Thursday, May 09, 2013 3:13 PM
To: Asterisk Users
I'm wondering if the SIP header join, RFC 3911, is supported in the
asterisk stack?
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EUS Networks
Office: 212-624-5943
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Bruce Reeves wrote:
I'm wondering if the SIP header join, RFC 3911, is supported in the
asterisk stack?
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Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
/priv. Is there a way that i can
detect it first so it does not try to dial it on the local before askng
dundi? thank you
regards,
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL
://lists.digium.com/mailman/listinfo/asterisk-users
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i
only see one extension there. even though i know i have 4 ip phones
registered to the asterisk.
TIA
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users
visit:
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== Spawn extension (from-sip, 400, 4) exited non-zero on 'SIP/156-08274b60'
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, April 23, 2008 4:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
It is not the dip peer that is failing
PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only
-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Tuesday, April 22, 2008 6:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Did you get this working?
On Thu, Apr 17, 2008
.
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Is there anything in the CLI about the sip peer? Can you show the
settings you have in sip.conf and the phone setting you entered?
Bruce Reeves
On Nov 16, 2007 9:00 AM, Jarga Jallow [EMAIL PROTECTED] wrote:
I am having trouble configuring my Polycom 320 IP phone. When I dial an
extension
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is, does the IP address on the phone change
when moving from on access point to another?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Wednesday, October 10, 2007 9:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
/mailman/listinfo/asterisk-users
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get is very small: only a handful of the
large body of Asterisk users actually naswered it.
So I am answering as someone using 1.4.
Bruce Reeves
Nortex Networks
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/Amerigo_Vespucci's name, *Americus
Vespucius*, in its feminine form, *America*, as the other continents all
have Latin feminine names.
Wikipedia and a memory for things historical.
Have a good 4th.
Bruce Reeves
Nortex Networks
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button blinking.
Any ideas?
Thanks,
James
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http
There are nine sites, 10 servers. While it is not a huge deployment by
some standards, it was simplified with DUNDi.
On 5/9/07, Olivier [EMAIL PROTECTED] wrote:
Just the sake of curiosity, how many sites (or user) did you interconnect
using DUNDi ?
Regards
2007/5/9, Bruce Reeves [EMAIL
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] wrote:
Hi Bruce,
Can you suggest me any documentation about using IAX truking?
Thank you.
Ronaldo.
Bruce Reeves wrote:
Yes it is.
On 5/3/07, *Ronaldo* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi all,
Is it possible to have something like this:
SoftPhone -(SIP
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a better way?
Thanks
Cameron
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to bypass putting another dhcp server on my network.
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that are vague and broad. Ultimately I think Verizon will suffer
from the court decisions more then anyone else, the true nature of their
leadership is not to satisfy the customer.
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the email addresses of
the participants? There is no way how to configure participants to the
conference.
I have seen this with my setup, I am using the client mode for emails, when
using firefox. Strange enough IE works. Most of our users are on IE so I
have not researched the why.
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the email addresses of
the participants? There is no way how to configure participants to the
conference.
I have seen this with my setup, I am using the client mode for emails, when
using firefox. Strange enough IE works. Most of our users are on IE so I
have not researched the why.
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of the problem?
;) Happy Easter!
Philipp
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a similar feature?
Thanks,
Chris
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a similar feature?
Thanks,
Chris
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, what is reason 104??
On 3/26/07, Jay Milk [EMAIL PROTECTED] wrote:
Bruce Reeves wrote:
I ran into a problem today while trying to compile chan_cellphone
version 17 on a CentOS 4.4 machine. Apparently the bluez and autoconf
versions were to old and as I tried to install the latest version, I
this
chan working on CentOS 4? If so can you help me through this?
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Yes, I recently saw this with a 501, in my case the network drop was
the problem. If you have a good tester then run it on the connection.
I had another drop near by and just swicthed to it.
On 3/21/07, Louis-David Mitterrand
[EMAIL PROTECTED] wrote:
Hi,
At one location we have a user whose
/07, Louis-David Mitterrand [EMAIL PROTECTED]
wrote:
On Wed, Mar 21, 2007 at 05:53:27AM -0500, Bruce Reeves wrote:
Yes, I recently saw this with a 501, in my case the network drop was
the problem. If you have a good tester then run it on the connection.
I had another drop near by and just
I have seen this product before and wondered, has anyone connected this to
Asterisk?
http://www.i-2-eye.com/index.html
As far as that goes has anyone seen a set top box video phones that work
with Asterisk?
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Have you looked at the Follwo-me feature in 1.4? It can require the
answering channel to accept the call. You might take a look.
On 3/16/07, Ritesh Agrawal [EMAIL PROTECTED] wrote:
Hi Folks,
I want to setup a follow me routine so that asterisk can call me on the
multiple numbers.
I tried some
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Sno Falls Credit Union
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Monday, March 12, 2007 9:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
That should be provided by your telco, if your referring to a PRI on a
Sangoma T-1 card.
On 3/15/07, Rob Vinson [EMAIL PROTECTED] wrote:
Does anyone know if I can get Incoming caller id name and number on a
sagnoma PRI.
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Cory,
Are the Polycom phones able to detect that the headset is off-hook? We
have had problems with 2 different brands of headsets working fine but
the phones seem unaware of the headset's hook state.
On 3/14/07, Cory Andrews [EMAIL PROTECTED] wrote:
I like any of the Polycom Soundpoint IP
Does SIPAddHeader(Alert-Info:) not do it?
On 3/12/07, Nikhil Jogia [EMAIL PROTECTED] wrote:
Hi All
I have just upgraded from Asterisk 1.2 to 1.4 and am having trouble with
with one of my ATAs not ringing.
Basically, when I execute the Dial command, an error occurs: Got SIP
response 400 In
Brandon
Your on the right track with what is can do. It will also be good to
look into what kind of QOS you can do on the T-1 connections between
offices. I have an 8 office setup similar to this and many of your
goals I have achieved and would be glad to offer ideas and such if you
want to
, Bruce Reeves [EMAIL PROTECTED] wrote:
Brandon
Your on the right track with what is can do. It will also be good to
look into what kind of QOS you can do on the T-1 connections between
offices. I have an 8 office setup similar to this and many of your
goals I have achieved and would be glad
I just completed a deployment of 8 sites connected via MPLS, and I
chose to go with the local * servers option and Sangoma hardware at
each site. I then put dundi in place to route calls between sites and
will later look at adding LCR. I'm with Steve on the cards, don't
skimp on cards or even
Steve,
If you can get this to work with your own choice of softphone please
post back to the list. I've wondered about it myself.
On 3/7/07, Steve Totaro [EMAIL PROTECTED] wrote:
It would be cool to get one of these and see if it can be hacked and
loaded with your favorite SIP or IAX softphone.
the displaysystemname=yes is the missing setting, I seem to remeber
not getting anywhere till I added that.
On 3/5/07, David Thomas [EMAIL PROTECTED] wrote:
On 3/2/07, Bruce Reeves [EMAIL PROTECTED] wrote:
Try renaming you column in the peers table to regserver
Thanks for the suggestion Bruce
Or the fact that www.virtualphoneline.com is part of DIDXchange and of
course you love it since you work for supertec.com, didxchange.com,
and virtualphoneline.com
On 3/5/07, Singer Wang [EMAIL PROTECTED] wrote:
Follow this link
Chris,
Here is how I might use this, I have a context called inside, is where each
of my extensions is dialed from. On my home box it looks like this.
[inside]
exten = 1000,1,Dial(SIP/1000,20,t)
What I would probably do is add the Notify command to each of my extensions
before my Dial, like so
Try renaming you column in the peers table to regserver
On 3/2/07, David Thomas [EMAIL PROTECTED] wrote:
I am trying to have asterisk update the system name in my realtime
peers, but it does not seem to be working. Here is what I've done so
far.
- added systemname = mysystemname in
What do you have setup in the res_mysql.conf file and extconfig.conf files?
Have you installed the asterisk addons for 1.4 to get support for mysql?
On 3/1/07, Mike Hammett [EMAIL PROTECTED] wrote:
queue show Show status of a specified queue
realtime load Used to
If re-invites are allowed then once both IAX endpoints are connected to
Asterisk and the call is active the server will attempt to step out of the
call. This actually works for both sip and IAX.
On 2/27/07, Joseph [EMAIL PROTECTED] wrote:
I find IAX connection with FWD very unreliable so I
In my experience having ap's with the same SSID and 3 channels of separation
overlapping worked if the phone could roam.
On 2/14/07, Ronald Wiplinger [EMAIL PROTECTED] wrote:
Can anybody tell me how I can set-up multiple access points with
overlapping coverage, so that a moving WiFi phone user
I have seen this when I have restarted the server from the asterisk CLI and
not a service asterisk restart command. I'm not sure as to why, but I always
assumed it had to do with the safe_asterisk file.
On 2/12/07, Earle Clubb [EMAIL PROTECTED] wrote:
Lacy Moore - Aspendora wrote:
I'm
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