Hey gang,
I'm hoping someone can help me out here. I've just noticed that on
two of my five Asterisk boxes (CentOS 4.4, Asterisk 1.2.12.1), I'm
getting the following translation cost for g729:
asterisk*CLI show translation
Server 1: g729 -26252525252426
On 06/11/2006, at 8:53 AM, Julian J. M. wrote:
Try forcing asterisk recalculate those costs:
Ok, that fixed it. Thanks! :)
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2/340 Gore StreetT: +61 (0) 3 9235 5400
Fitzroy, VIC
On 27/10/2006, at 7:22 AM, Thomas Winter wrote:
I have to set up an Asterisk with an 4-port BRI card.
Hylafax should send and receive fax.
Will this work reliable?
I have a Eicon V-4BRI (which is in fact a voice-only board) that does
faxing via HylaFax/IAXmodem and its flawless. However,
On 23/10/2006, at 10:13 AM, Joseph wrote:
I'm trying to log-in externally (from PSTN line) to check my
voice-mail so I created context to authenticate log-in
Just create an inbound route to VoiceMailMain(). Then, press *
during the outbound message and it'll prompt you for a password. Hey
On 23/10/2006, at 2:24 PM, Martin Joseph wrote:
It doesn't work. pressing * during my outgoing message does nothing.
Works for me. 1.2.12.1 with FreePBX. When I press *, I get a
password prompt. Entering my password gets me into the main
voicemail menu.
cYa,
Avi
--
National Manager -
On 23/10/2006, at 2:26 PM, Eric ManxPower Wieling wrote:
The previous poster is obviously running some Asterisk GUI.
Yes, sorry. I am running FreePBX, but I didn't notice the | in the
call to VoiceMailMain, otherwise I would've mentioned it. :(
My bad.
--
National Manager - Special
On 23/10/2006, at 2:35 PM, Eric ManxPower Wieling wrote:
Works for me. 1.2.12.1 with FreePBX. When I press *, I get a
password prompt. Entering my password gets me into the main
voicemail menu.
FreePBX is NOT Asterisk.
Yes, I know that. Hence the 1.2.12.1 *with* FreePBX statement. I.E.
On 16/10/2006, at 2:32 PM, Paul Hales wrote:
We are currently writing a reception console for Asterisk - if
anyone is
interested in beta testing it, feel free to ask.
If it can handle multiple Asterisk servers -- ME, ME! PICK ME! PICK
ME! :)
Thanks,
Avi
--
National Manager - Special
On 10/10/2006, at 2:10 AM, Noah Miller wrote:
Quite right. I'm blaming the inadequacies of my OS on vsftpd. vsftpd
just uses your OS user accounts. On the Tao linux box that I had it
installed on, you couldn't do capitals in user account names. My bad.
Which is weird, because I thought
On 08/10/2006, at 3:00 PM, Dean Collins wrote:
Whats the best ftp server to upload Polycom phone cfg’s from? I’m
finding it a bit hit and miss using BTF server.
I'm using vsftpd quite successfully on several Asterisk boxes with
Polycom IP501 phones. Though, I'm now considering switching
On 09/10/2006, at 5:07 AM, Noah Miller wrote:
username and password is PlcmSpIp. vsftpd cannot handle capitalized
usernames, so if you want to use vsftpd, you have to manually
re-configure the username on each phone.
I use vsftpd and I'm using the default PlcmSpIp username just
fine. :)
On 08/10/2006, at 9:34 PM, Paul Hales wrote:
I have seen an Optus SHDSL box set up incorrectly before - and the
tech
re-visited and set it up correctly within hours of being informed.
Same with my Optus SHDSL box: The first tech misconfigured, so I kept
getting PRI restarts on my Sangoma
On Mon, October 9, 2006 11:46 am, Dean Collins said:
Are you able to track real time from a windows machine the transactions
occurring on your asterisk server if you have vsftpf installed?
Yes... In an SSH session, tail -f /var/log/vsftpd.log will show you
everything you need.
Also, I have
On 09/10/2006, at 12:12 PM, Dean Collins wrote:
can anyone give me an idea on how this reboot script works?
I actually just use the SIP notify command on the Asterisk console to
remotely reboot my Polycom phones. It requires a pre-configured
sip_notify.conf file and the Polycom option to
On 05/10/2006, at 4:25 PM, Abdul wrote:
But i am little confiuse why i am not able to install MySQL Real-
Time. here is the Error when i am trying to make all for asterisk-
addons-1.2.4.
You need to install the mysql-devel package to get the header files.
cYa,
Avi
--
National Manager -
On 04/10/2006, at 1:55 AM, Matt wrote:
How can I make * aware of the other ext on the remote box so the
DID caller can access them like he can with the local box?
On each box, define the other range:
Box A:
exten = _9XX,1,Dial(IAX2/BoxB/${EXTEN})
Box B:
exten =
joea, j4computers wrote:
So, now I am struggling with a Suse SLES 9 install, that seems reluctant to
co-operate.
I have a number of boxes running CentOS 4.4 with Asterisk 1.2 and
FreePBX: Because I install everything manually, I know it all works,
without the overhead of the Trixbox features
Mike Dent wrote:
I'm curious how you installed Trixbox?
There is a tar.gz version of Trixbox that can be installed over an
existing RHEL4 or CentOS installation.
However, removing Trixbox is very difficult. You are better off
reinstalling RHEL4 and then installating Asterisk from scratch.
Hi guys,
I have what is probably a very noob question. I've tried to search the
wiki, but my lack of knowledge is hindering me in finding the right
keywords:
I'd like to know what the packet size of an IAX2 packet is, if its using
the ilbc codec.
Now I'll tell you why, so you can tell me
Brent Franks wrote:
We ran into the same thing, and the only way I can get it to work
(which is goofy, but it does work) is modprobing the same device
multiple times.
Try waiting after modprobe zaptel for udev to create the device nodes. I
do this:
modprobe zaptel
wait 5
modprobe wctdm
Roland wrote:
I've tried all those at voip.info.org but I just couldn't get it
right. and I don't have the luxury of time to try figure out how to
make it work by myself.
The official FreePBX install docs (which have Asterisk instructions as
well) for CentOS are here:
Kevin P. Fleming wrote:
if could download SVN branch-1.2 and try it out on your system to see if it
solves your issue.
Is there a Wiki page or similar describing how to checkout SVN for
Asterisk? Also, will I need to checkout and compile SVN versions of
Zaptel/Libpri/Addons (as I use all
Avi Miller wrote:
Is there a Wiki page or similar describing how to checkout SVN for
Asterisk? Also, will I need to checkout and compile SVN versions of
Zaptel/Libpri/Addons (as I use all three)?
Replying to myself to say that I've found Digium's instructions and I'm
testing SVN on my test
Avi Miller wrote:
Replying to myself to say that I've found Digium's instructions and I'm
testing SVN on my test server now. :)
And again to say that it seems work just fine with the SVN code. Thanks
Kevin!
--
National Manager - Special Projects
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Hey guys,
I've been trying to change my Asterisk setups to use canreinvite=yes.
I'm having a small problem with my Polycom IP501 phones and transferring
calls.
If a call comes in via my ISDN BRI lines (using chan-capi), I can
successfully transfer the call using the Polycom Blind Transfer
Hey guys,
I need some assistance in tracking down the cause of audio problems that
are occurring at two of my sites:
Both sites run Asterisk 1.2.10 and use Sangoma A101u PRI cards. Both
sites are reporting that audio in calls is dropping out during words,
so that the other caller (i.e. the
Avi Miller wrote:
Does anyone have any suggestions on where to look next? My users are
getting increasingly annoyed and I'm quickly running out of ideas.
Replying to myself to note that this is now happening on outbound calls
via ISDN, i.e. calls that don't use IAX2 or the inter-office
On Mon, August 28, 2006 5:17 pm, Erik said:
Through what means are both sides connected, 1:1 xDSL?
All offices are connected via 512/512 SDSL.
What bandwidth, are you using tunnels (pptp/gre/ipsec), how many
concurrent calls etc.
No tunnels (that I'm aware of). Very few concurrent calls,
On Mon, August 28, 2006 5:21 pm, Matt Riddell (IT) said:
Are you using realtime?
No, the Asterisk boxes are managed by FreePBX which creates .conf files. I
have two boxes playing up (the ones with PRI connections). My other three
servers that use BRI are just fine. Calls between the other three
On Mon, August 28, 2006 8:10 pm, Rich Adamson said:
Is this a new installation, or, were the boxes working okay for a while
and they just now started having problems?
Its not a new installation: Calls have been fine for at least a month on
one server and about 4 months on another. Both servers
DM wrote:
Why do you think the problem may be with the FTP server? I've been
running vsftpd on several different systems, all with Polycom's.
There were reports that the Polycoms preferred some FTP servers over
others, but I also use vsftpd (using the default PlcmsSpIp
username/password
Hey guys,
Last week I changed my queues from using proper agents and
AgentCallbackLogin() to using the the FreePBX default with fixed agents
(which uses the Local/[EMAIL PROTECTED] style for the member= field).
I've also upgraded to Asterisk 1.2.10 and FreePBX 2.2.0 Beta 1.
Since then, I
On Fri, August 11, 2006 4:26 pm, Wolfgang Paul Rauchholz said:
allow=g729
allow=g723
Do you have the g729 and g723 codecs installed? They are not installed
with Asterisk by default.
cYa,
Avi
___
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hugolivude wrote:
I'm able to get the number to change but the name is always Unknown
Name. I've tried numerous combinations of quotes, but just cannot
get the name...
I use Caller Name401
Note, no space between the closing and the character. Seems to work
for me and Polycom phones.
Dean Collins wrote:
Yep, but didn’t [EMAIL PROTECTED] have a folder to store these files on?
Does freepbx?
You mean TrixBox? I know they're working on a phone provisioning system,
but I thought it was just for Cisco and Grandstreams. Check with the
TrixBox guys at http://www.trixbox.org
Stephen Murphy wrote:
Can you simply replace your current sip.Id and sip.ver files with the latest
firware files or is this dangerous?
That's what I did, after doing a diff of the old and new original
sip.cfg and phone1.cfg files to make sure there weren't any major
changes/additions.
--
Stephen Murphy wrote:
And this worked without issues?
It did for me. YMMV, depending on the changes you made to your sip.cfg
and phone1.cfg. :)
--
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2/340 Gore Street T: 1 3000 SQUIZ (77849)
Stefan-Michael. Guenther (in-put GbR) wrote:
I have a fax server with an AVM Fritzcard that is connected to port number 4
of an EICON DIVA Server 4 BRI.
If the inbound is always going to be fax, set faxdetect=off in
capi.conf, so that it just runs the default.
Otherwise, add a fax
Andrea Spadaccini wrote:
Is there any hope to change the caller-id on a BRI line?
I can change my Caller ID on my BRI lines to anything within my DID range.
Hope that helps,
Avi
___
--Bandwidth and Colocation provided by Easynews.com --
Douglas Garstang wrote:
The phone will quietly not be able to write to the contacts directory.
However, it seems the directory on the phone is maintained. I still
can't work out how to get the Polycoms to replace any locally added
directory items with a master list from the provisioning
Stephen Murphy wrote:
My question is: How do I get the
current config files the phone is using off the phone?
AFAIK, you can't. :( You can only provide new configuration files from
your FTP/TFTP server. However, the Polycoms do strange things when
they've been configured in multiple
root linux wrote:
I am having problem sending out fax from fax using an
ATA connected to the asterisk.
Your system is detecting the fax and trying to receive it at the same
time. I had the same problem for a while, and Armin nicely changed
chan_capi for me. :)
Essentially, if you're using
root linux wrote:
I am not using any Zaptel card... I am doing a
back-to-back to Verso C5CM via Internet
Wow. You're going to probably run into problems trying to fax over a
VoIP connection. Other people can explain why far better than I can. :)
--
National Manager - Special Projects
Douglas Garstang wrote:
Yes, we tried to do the same thing. We wanted our Asterisk system to be multi-homed.
My head office Asterisk box is multi-homed: I have three networks across
two NICs. One dedicated to hardphones, another to the local LAN (and
PC-based softphones). The third network
Hey guys,
Could someone recommend some good hardware echo cancellation devices for
a single ISDN E1 line? I need something to sit between the wall and a
Sangoma A101u PCI card.
Preferably a device that I can source in Australia! :)
Thanks,
Avi
--
National Manager - Special Projects
Doug Lytle wrote:
A Tellabs 2572 64ms EC. Check ebaY. Instructions on the Wiki.
Anything that requires a little less soldering? :) I was hoping for a
boxed solution.
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National Manager - Special Projects
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2/340 Gore Street T: 1
Bill Gibbs wrote:
Any ideas or suggestions? Just trying to keep the number of button
presses to a minimum.
The number of button presses would be the same though:
1. Pick up the phone, dial 7 3 0 0 (four buttons)
2. Hit line 3, dial 3 0 0 (four buttons)
You could configure the line 3 button
Tom wrote:
I don't like the fact that CentOS is nothing more than a copy of RH
Enterprise Server. It is almost like running a Windows clone. I would
rather find and run something better.
While I love CentOS for the very same reason: I get all the benefits of
Red Hat Enterprise Linux
Douglas Garstang wrote:
I don't think unison is a workable solution. It doesn't scale. The network and
system load would increase exponentially as we added asterisk servers to our
cluster.
If you're clustering that many boxes, I'd investigate fibre channel SAN
and GFS. That way, each node
Khaled Chehab wrote:
Hi I make a Digital Receptionist ,but how can I attach it to an extension
[EMAIL PROTECTED] is now called TrixBox. You'll get a lot more support at
http://www.trixbox.org
cYa,
Avi
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National Manager - Special Projects
Melbourne / Sydney / Canberra / Hobart / London
Tyler Retzlaff wrote:
to use Active card(s) and I need to accommodate 2 x BRI TE/PTP
services.
I'm devoted to Eicon Diva 4-BRI cards: They're expensive, but they have
onboard DSPs and Echo Cancellation, which is awesome. Also, great Linux
driver support and chan_capi-cm support for
Johnny Stork wrote:
CentOS behind trixbox is a relatively complete CentOS system?
The installation of CentOS is sufficient to support TrixBox, but you can
always add additional packages using yum.
cYa,
Avi
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National Manager - Special Projects
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Ronald Wiplinger wrote:
What do I miss ???
Your current blind transfer setting is ##, so try ## 632 instead.
cYa,
Avi
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National Manager - Special Projects
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2/340 Gore StreetT: +61 (0) 3 9486 0411
Fitzroy, VIC F: +61
Stephen Bosch wrote:
All I want to do is get my Polycom 501 to register with a working
Asterisk server. I want to do the configuration locally on the phone
through the console.
The console is very tedious. Why not use the web interface instead? Let
the phone get an IP address via DHCP and
Kerry Garrison wrote:
There are several listed at http://voip-info.org. For Management check out
FreePBX, for recorded calls look for Asterisk Recording Interface.
FreePBX includes ARI, btw. :)
cYa,
Avi
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National Manager - Special Projects
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On 26/05/2006, at 7:49 PM, Jamie Heckford wrote:
Can anyone shed any light on this issue? I thought it could be
asterisk
is trying to Dial to soon so I added a Wait in the dialplan but it
didn't seem to work.
Polycoms have their own dialplan built into the phone. Depending on
how you
On 25/05/2006, at 8:14 PM, Bob Chiodini wrote:
message that says Asterisk mail then short pause then the word
mailbox then a very long pause, then a request for a password. I
Its asking you for your mailbox number at that point, then pausing to
allow you to enter the mailbox number. When
On 25/05/2006, at 8:57 PM, Bob Chiodini wrote:
I don't hear a request for my mailbox number. Should it say
something like Enter mailbox number?
I believe the prompt just goes Mailbox? -- its not great. But,
there's no other prompts being played in your output.
--
National Manager -
Mimmus wrote:
I'd like to drop this script: does anyone can explain me what is its main
job?
Dialparties.agi is used to test all of the submitted destinations for
Call-Waiting and Call-Forward settings before passing the final
extension(s) that can be called back to Asterisk.
--
National
Michael George wrote:
We are moving our asterisk 1.0 system to a new Asterisk @Home
system (2.8) and I am the one in charge of doing it.
You're probably better off asking at the FreePBX forums
(http://forums.freepbx.org).
In answer to your question: The default behaviour for MeetMe changed
On 22/05/2006, at 9:13 PM, [EMAIL PROTECTED] wrote:
Is it a way to record a welcome message and use it ?
Dial into VoiceMailMain() and hit 0 for Mailbox options. You can
record both an Unavailable and a Busy message. :)
cYa,
Avi
--
National Manager - Special Projects
Sydney /
On 18/05/2006, at 6:51 PM, Wayne Gemmell wrote:
are a good option (extensive, but come highly recomended from most
that
I hear). Good luck and happy hunting.
Ouch, you weren't joking. 1453 Euro!
But worth every penny, imo. I have a few servers running Eicon Diva
Server V-4BRI cards and
On 17/05/2006, at 7:36 PM, richard Coco wrote:
[local]
exten = 2001,1,Dial(SIP/2001,10,tr)
exten = 2002,1,Dial(SIP/2002,10,tr)
[notify]
exten = 2001,hint,SIP/2001
exten = 2002,hint,SIP/2002
Try this:
[local]
exten = 2001,1,Dial(SIP/2001,10,tr)
exten = 2001,hint,SIP/2001
exten =
On 17/05/2006, at 8:27 PM, richard Coco wrote:
unfortunately i still don't see subscribe request in
the sip debug trace.
Have you configured your phone to subscribe to the extension? :)
cYa,
Avi
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On 17/05/2006, at 1:26 PM, MBIT Technologies wrote:
I have mISDN installed and working correctly but I am unable to
receive a fax through the connection.
I have NVFaxDetect and RxFAX running on my CAPI channels, so I know
that works. :) Does NVFaxDetect detect the fax correct? Does it
On 17/05/2006, at 1:45 PM, MBIT Technologies wrote:
I can't see any fax detection at all in my call logs.
What does your dialplan look like for incoming calls? Do you give
NVFaxDetect enough time? I find that 4 seconds is good, but 2 seconds
is dodgy, for example.
cYa,
Avi
On 17/05/2006, at 2:29 PM, MBIT Technologies wrote:
This is what Im getting when I try to receive a fax
Yeah, looks like NVFaxDetect isn't dropping to the fax extension. You
may want to check with the NV guys to see if it works with mISDN. For
reference, I have it working with the Eicon
Michael J. Tubby B.Sc (Hons) G8TIC wrote:
call then transfers it on to another extension transferee (recipeient)
sees the Caller*ID
This behaviour changed in Asterisk 1.2 -- add o to your Dial options
and Asterisk will retain the original Caller ID on transfer.
--
National Manager - Special
On 11/05/2006, at 6:03 PM, Isaac Xiao wrote:
Would any one advice how implement Diva Server BRI or PRI card to
support fax and data modem? In Eicon’s website, it says that they
support these. But there is no FXS port on the card, how it can be
connected to Fax machine or data Modem?
It
Time Bandit wrote:
And the *_additional.conf files are the ones overwritten by the config
in the DB. So you can edit the other ones.
You could, but it'll get overwritten by any FreePBX upgrades. The *.conf
and *_additional.conf files are controlled by FreePBX and can be
overwritten. The
Avi Miller wrote:
You could, but it'll get overwritten by any FreePBX upgrades. The *.conf
and *_additional.conf files are controlled by FreePBX and can be
overwritten.
I thought I should clarify this statement: I meant that FreePBX could
overwrite both the *.conf and the *_additional.conf
Rich Adamson wrote:
Maybe its just me, but it appears its no where near usable even with the
latest beta1 code.
Its just you. I have FreePBX running on 6 production boxes across the
country. I do very little additional scripting. 5 of the servers have a
Eicon Diva Server V-4BRI card. The
Rich Adamson wrote:
address zap interfaces, but implies all four lines have to drop into the
same context. Not usable given the above.
The new beta (2.1) allows you to route inbound based on Zap channel --
you could set each channel to route to a specific destination, and
FreePBX will create
Rich Adamson wrote:
Well... all those things were installed with FreePBX, they just didn't
grow there. ;)
Honestly, those utilities never been part of FreePBX (nor are they
installed by FreePBX). They are only ever installed as part of
[EMAIL PROTECTED] However, one of the FreePBX developers
Rich Adamson wrote:
Actually, they were installed by FreePBX and I still have the iso disk
to prove it
The ISO is [EMAIL PROTECTED], not FreePBX. FreePBX has never shipped as an
ISO. FreePBX is simply one of the many software applications that have
been combined to form the [EMAIL PROTECTED]
Rich Adamson wrote:
So, how do you know which conf files one can hand edit versus those that
might be overwritten?
You may only change the *_custom.conf files. :)
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2/340 Gore Street T: +61 (0) 3
Rich Adamson wrote:
zap interface, but apparently undid what existed to edit conf files,
crm, etc. That made things look like a step backwards.
Yeah, a lot of people get confused about that. I was just trying to
clear things up. :)
--
National Manager - Special Projects
Sydney / Melbourne
Hey guys
This is probably for Armin, but I thought maybe someone else might have
something I could try. I'm having a problem with one of my Eicon Diva
V-4BRI cards and I'm trying to work out if its a driver configuration
error, card failure or telco problem:
I have an Asterisk box running
Avi Miller wrote:
This is probably for Armin, but I thought maybe someone else might have
something I could try. I'm having a problem with one of my Eicon Diva
V-4BRI cards and I'm trying to work out if its a driver configuration
error, card failure or telco problem:
Replying to myself
Olivier Krief wrote:
When writing receiving a fax over CAPI, do you mean receiving a fax
over CAPI with Asterisk and processing it with spandsp ?
No, with a full Eicon Diva 4BRI card, it does hardware faxing. Instead
of using rxfax (which uses spandsp), you'd use capicommand(receivefax)
Tielin Xu wrote:
I noticed that there is no ip address stored for my softphone in Mysql,
how does the Asterisk know which computer my softphone is running? I
checked the config files, no softphone registrations in sip.conf.
freePBX stores your phone information in sip_additional.conf and
Tielin Xu wrote:
I'd like to use FreePBX, it seems some setup inconsistency with
Asterisk RealTime, do you know any other good admin tool for Asterisk?
FreePBX is not designed to work with Asterisk RealTime. I don't know of
a GUI to configure RealTime myself. :)
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National Manager -
Jim Freeze wrote:
I have a TDM card with 4 lines on a hunt group coming in.
The latest version of FreePBX (2.1 Beta 1 - currently in SVN, but should
be released soon, I'm told) allows you to create inbound routes based on
Zap Channel, which I believe is what you're look for.
You may want
Armin Schindler wrote:
Using spandsp and V-4BRI does not work?
That will work. It's just that the on-board fax capabilities won't work, but
any other software fax will work like with other cards.
Just a note that I've never managed to get this to work on my V-4BRI
cards: If I attempt to
Kerry Garrison wrote:
Didn't help. Could I be missing something else?
My phone.cfg looks like this:
mwi
msg.mwi.1.subscribe=300
msg.mwi.1.callBackMode=contact
msg.mwi.1.callBack=*97/
And sip.conf for extension 300:
[300]
username=300
type=friend
secret=***
record_out=Adhoc
Armin Schindler wrote:
The configuration is as easy as with BRI lines. Can you provide more (like
your confs and verbose/debug output)?
Also (this isn't directed at you Armin, but I found your email to reply
off of to maintain the threading), I created a Wiki page over at the
freePBX
[EMAIL PROTECTED] wrote:
Asterisk says it has 30 capi channels available, but my mistake may be
in configuring the trunks...
When I was debugging my Eicon Diva 4-BRI board, I found it useful to
play with extensions_custom.conf (in AMP) just to ensure I got the
Custom Dial String absolutely
Jim Rice wrote:
I only asked this list as a last resort, having already exhausted many
other avenues. I even mentioned that it was OT, but have seen numerous
postings for phones of all kinds.
A thought: I had similar problems with one phone of mine after I
power-cycled it during the
Jim Rice wrote:
Anyone seen this before?
I'm not sure about that exact error, but I get these systems if I stuff
up the XML in sip.cfg or phone1.cfg (or the specifc phone equivalents).
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2/340 Gore
Thomas Broda wrote:
Which component do I have to install in order to get the Maintenance
setup?
The Maintenance tab is part of [EMAIL PROTECTED] and not AMP/freePBX. You'll
only see it on an [EMAIL PROTECTED] installation.
cYa,
Avi
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National Manager - Special Projects
Melbourne /
Noah Miller wrote:
Another idea: Can you create the mac address-directory.xml files as
symlinks to the central file?
Great idea and it works, too! :)
Now I just need to make 50 symlinks.. luckily I have a list of mac
addresses, so its just a Bash script away.
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National Manager -
Dinesh Nair wrote:
the symptoms are that calls from a SIP client to NetMeeting rings on
NetMeeting, but upon answering the call in NetMeeting, no audio is passed
between the two. eventually, the call times out and hangs up.
I had a similar problem connecting Asterisk to an Avaya IP403 via
Dinesh Nair wrote:
more tests reveal that with ohphone, calls from SIP-ohphone work fine
with audio passed both ways. however when ohphone calls a SIP device,
the call is hungup when the SIP device answers.
This was sort of my problem too. I have two Asterisk servers, with an
IAX2 trunk
Giuseppe wrote:
I'm always getting this error when echo cancellation should start.
What does your /etc/asterisk/capi.conf look like?
Also, have you configured your Eicon correctly? You may need to enable
the Eicon web interface and check that each port is correct.
--
National Manager -
Giuseppe wrote:
ntmode=yes ;if isdn card operates in nt mode, set this to
This should be set to no -- you should be in TE mode.
echotail=64 ;echo cancel tail setting
bridge=yes ;native bridging (CAPI line interconnect) if
I don't have either of these
Mojo with Horan Company, LLC wrote:
if you reboot your phones from the asterisk server ie via cron or so,
that reboot script could potentially delete the phone-specific directory
xml before the sip message is sent
Sadly, that doesn't work -- the Polycoms store their directories locally
as
Giuseppe wrote:
Can anybody tell me if there is some error or something missing in this
configuration please?
I have the same card in a few of my servers and the echo canceller works
just fine. I'm not 100% sure, but something does jump out at me:
== ISDN3: Answering for 'x'
--
Hi gang,
I know this is off-topic for Asterisk, but I don't know where else to
ask: I've setup a central directory.xml file for my Polycom IP501 phones
with a list of all the internal extensions. None of them have sd1/sd
as I don't want to enable any speed dials, just have a list in each
Doug Lytle wrote:
shameless plug Something like this perhaps?
VERY cool!
I agree. When does that get released? :)
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National Manager - Special Projects
Melbourne / Sydney / Canberra / Hobart / London /
2/340 Gore Street T: +61 (0) 3 9486 0411
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Jim Hanlon wrote:
1. The alterations to the config files made via AMP Setup pages are archived
in the Asterisk DBMS, but changes made via the AMP
Maintenance pages are not (Apparently. It's hard to be sure what the rules are).
This is an [EMAIL PROTECTED] issue: The Setup page is provided
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