Displaying 20 results from an estimated 5000 matches similar to: "AP/OR Manager"
2004 Aug 24
0
Using Lucent/Avaya 64XX sets with asterisk
Hi all,
I am new to asterisk, but experienced with Linux and Avaya/Lucent G3R
hardware/software. We currently have a large inventory of spare Lucent
64XX phone sets. I'm considering setting up an asterisk system for a
remote office, and would like to use the Lucent sets because they are
cheap, and familiar to my end users. Does anyone know of a card that
can drive these sets? I think
2001 Mar 16
1
login from another user
LS,
I've create on the SCO unix the same loginname as on my winNT pc
sharing a directory is no problem I have read and write access to it.
The problem arise when I try to login from another winNT account let say
yoyo
then a pop-up windows appears. When I fill my origineel account with
password
I am refuse. Does someone know how to solve this?
dump of my smb.conf file:
[global]
2005 Aug 04
0
h323 CALL PROBLEM TO / FROM AVAYA(UCENT)inity
May be force be with you
hello,
We want to make H323 calls beteen asterisk and avaya(lucent) pbx.
We create node-name,H.323 signaling group,trunk,
but we can not make H.323 calls to asterisk. Also no warnings exist in
debug.
Instead of giving the IP of Asterisk ,i give my computer's IP and run
SJPhone ith H.323 GUI.
In this time, connection is established.
SJPhone accepts H323 calls but
2005 Aug 04
0
h.323 Call problem asterisk to\from lucent(avaya) definity
Hello,
We want to make H323 calls between asterisk and avaya(lucent) pbx.
We create node-name,H.323 signaling group,trunk,
but we can not make H.323 calls to asterisk. Also no warnings exist in
debug.
Instead of giving the IP of Asterisk ,i give my computer's IP and run
SJPhone ith H.323 GUI.
In this time, connection is established.
SJPhone accepts H323 calls but Asterisk does not.
Do
2008 Jun 18
1
Avaya IPSoftPhone
Has anyone had any luck getting the Avaya IP SoftPhone to work in wine?
I get these errors on Ubuntu 8
Code:
preloader: Warning: failed to reserve range 00000000-60000000
wine: creating configuration directory '/home/microchp/.wine'...
preloader: Warning: failed to reserve range 00000000-60000000
preloader: Warning: failed to reserve range 00000000-60000000
err:dosmem:setup_dos_mem
2013 Jun 08
0
H.323 Trunk between Asterisk 11 and Avaya
Hello,
I'm trying to create a H.323 trunk between Asterisk 11 and Avaya. I have
done this before between Asterisk 1.6 and Avaya but had some issues placing
external calls from the Asterisk to the Public network which is connected
to Avaya. I'm trying to create that trunk on Asterisk 11 because the 1.6 is
outdated and has no support.
On the Asterisk side I have Aastra 6731i SIP phones
2009 Jan 30
2
Asterisk with Avaya
Hi !
I am trying to connect Asterisk with Avaya Definity.
I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html
The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything
Example
Asterisk ---> Avaya
--
2007 Jun 26
0
[1082] trunk/wxruby2/doc/textile/hyperlinkctrl.txtl: Added HyperlinkCtrl and HyperlinkEvent documentation
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.1//EN"
"http://www.w3.org/TR/xhtml11/DTD/xhtml11.dtd">
<html xmlns="http://www.w3.org/1999/xhtml">
<head><meta http-equiv="content-type" content="text/html; charset=utf-8" /><style type="text/css"><!--
#msg dl { border: 1px #006 solid; background: #369; padding:
2006 Feb 14
0
Lucent Avaya Partner ACS T1 module
I'm trying to connect an Asterisk system to an Avaya Partner ACS R6
system. The problem I'm having is that I cannot get the partner system
to get CallerID over the T1 modlue. The partner is using the T1 with E
& M signalling (which I don't think can be changed), and whatever I
tried didn't work. My only option right now is to get FXS ports on the
Avaya side plugged into the
2007 Sep 12
2
Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?
Hey all!
I'm newbie in the Asterisk World but old in other telephony systems like
Lucent/Avaya, Sopho, Siemens and Linux/Unix system.
I'm in doubt, as based system, should I install Fedora, Debian,
Slackware, FreeBSD our Sun Solaris? Which is more robust for a small
Asterisk system, about 8 extensions, 4 hardphone and 4 softphone?
Thanks in advance!
Euler
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2006 Jan 12
0
Transfer issue with a Cisco CCM/phone
Hello,
We have a mixed environment here consisting of a number of Avaya PBX
systems, a group of Cisco Call Managers, an H.323 gateway on a Cisco
router, and an Asterisk server. The PBX land is connected to the VoIP
land using the Cisco router/H.323 gateway.
The Asterisk system is running code from the CVS tree from around mid
Oct 2005. The OH323 driver on the system is from inaccessnetworks,
2015 May 14
1
chan_ooh323 to sip , no connected line info
Hello!
We have asterisk connected over PRI no our phone network, so I'm avaya
PBX user.
Asterisk connects to another avaya system over h323.
Connection can be shown as
avaya--PRI-asterisk--h323-avaya
When I do call as avaya user I see name of remote end avay user,
i.e. connected line info.
As I see in debug remote side send is as
14:07:29:758 Received H.2250 Message = {
14:07:29:758
2006 Jan 12
0
Re: Transfer issue with a Cisco CCM/phone (Peckham, Christopher)
Christopher-
Nothing like defining a complicated environment. I do have some experience
in this arena- but unfortunately, not with the OH323 driver- I generally
stick to the Nufone driver, as I find it more reliable overall. YMMV. One
thing that might help is if you could tell us if it ever worked, or if this
is a new problem that's cropped up since a particular change.
Still- there are
2004 Apr 29
1
User picks up phone, hears another call, not dialtone
First of all - Many, many thanks to Mark for his troubleshooting and fix
of bug 1320 (FXO_KS signalled Zap Channels on Adtran 750 Channel Bank
Stuck in Rsrvd State).
I have heard complaints that once every couple weeks, when a user picks up
their analog phone (t1 span off of a TE410P into an Adtran 750 with 6 FXS
cards), they don't get dialtone, but instead, hear another conversation.
2008 Nov 07
0
asterisk - avaya ip office SIP trunking
Hi * Users,
I ran into a problem when I was trying to communicate an avaya IP Office
talk to asterisk with SIP Trunking. I had successful calls from asterisk to
Avaya but not from avaya to asterisk.
Can someone provide me insight on how to address it or the path to resolve
it.
The error I get is mentioned below: (dialing 32564 from avaya to asterisk)
"[Nov 6 17:14:23]
2008 Nov 07
1
Help with asterisk and avaya SIP trunking
Hi * Users,
I ran into a problem when I was trying to communicate an avaya IP Office
talk to asterisk with SIP Trunking. I had successful calls from asterisk to
Avaya but not from avaya to asterisk.
Can someone provide me insight on how to address it or the path to resolve
it.
The error I get is mentioned below: (dialing 32564 from avaya to asterisk)
"[Nov 6 17:14:23] WARNING[6227]:
2008 Apr 30
0
AVAYA 8300 integration with asterisk 1.2.x
Hi All,
I need help with integrating AVAYA 8300, the avaya can do outbound
calls but cannot do inbound calls, im sending calls from sip to avaya
using E1 ISDN line. My config was based on aspect dialer it's working
with aspect but not with avaya.
My config and error is below.
zaptel.conf
span=1,1,1,ccs,hdb3
bchan=1-15,17-31
dchan=16
zapata.conf
group=0
context=avaya
switchtype=euroisdn
2004 Aug 11
2
Avaya and Asterisk
So far I have not found a way that I can register the Avaya phone
with Asterisk. From what I have found so far is that Avaya phone
needs the Avaya Media Server and Avaya Gateway.
Looking at the h.323.conf (in Asterisk) and the file 46xxsettings.txt
(avaya file located in tftpboot) there are no settings to make the
phone initialize.
I have sent an email to the Asterisk Users Mailing List to see
2010 Jun 16
0
H323 Trunk Problem calling from Asterisk to Avaya PBX
On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi <shinacalypse at gmail.com>wrote:
> Hi!
> I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide
> a conference bridge for an existing Avaya PBX. I have no control over the
> Avaya system, but I am able to speak with the admin in charge when I need
> stuff done. I am running all this in a VirtualBox
2009 Jan 22
1
Help with Avaya integration
Hi,
I'm trying to integrate my asterisk PBX to an Avaya PBX. I am using
chan_ooh323 from asterisk-addons.
I am able to make a call from SIP Phone -> Asterisk -> Avaya -> Station
(phone) and vice versa.
I am also able to make a call from SIP Phone -> Asterisk -> Avaya -> PSTN.
However I face problems when I make DID calls from the PSTN. The DID
calls are made through