Markos Vakondios mvakondios at gmail.com writes:
Hi,
I use a simple scheme:
SIP video phone A (h264/Asterisk 1.8.11) ---IAX2 trunk SIP video
phone B (h264/Asterisk 11.7.0)
When calls from A to B and vice versa drop on pickup.
On B side:
[Oct 24 16:33:49]
on my cisco phones :)
Interestingly - I've got another query - but will post another question
when I've had chance to play more.
Cheers
Wayne.
Wayne wrote:
Hi all,
I've just built a new installation of CentOS release 5.3 (Final) and
have installed both
http://lists.digium.com/mailman
which the missus has a moan at every so often :-) ).
One thing I am unsure of - how do I get the dumps / information you
want in a suitable format.I'm still a novice with Linux / Asterisk but
I'll gladly get anything to help out (just need some pointers in the
right direction).
Cheers
Wayne
Sorry to bump my own message - but had a mail server problem so don't
know if I missed any replys :(
Ta
Wayne.
Wayne wrote:
Hi all,
I've just built a new installation of CentOS release 5.3 (Final) and
have installed both
http://downloads.asterisk.org/pub/telephony/asterisk/releases
.
Still open to an suggestions though :)
Thanks
Wayne.
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a 1.2 setup for years (using
chan_skinny?) but thought it time to update Asterisk.
Anyone have any pointers please on what to check next?
Thanks,
Wayne
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impress as
much as it should, so interesting days ahead :))
Thanks again
Wayne.
Sigma Networks wrote:
I have a production PBX (1.6.0.9) 150+ phones with MS Exchange 2007
and OCS working very well out of the box. We're using SIP/TCP support
in 1.6.x; Believe it or not the most challenging
.
Thanks
Wayne.
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be happening in later versions but all
seem to be dated quite old now and nothing suggesting that its now done.
Thanks
Wayne.
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- still getting there - but Asterisk
'seems' to be running - albeit VERY idle at this point!
Cheers
Wayne.
Dana Harding wrote:
Hello Daniel,
You will find the information at http://www.voip-info.org/ and
http://oreilly.com/catalog/9780596510480/ (.PDF downloadable from the
Online Book link
the day.
I shall take another look at both options.
Thank you
Wayne.
Michiel van Baak wrote:
On 08:26, Fri 10 Oct 08, David Gibbons wrote:
You need to check out the chan_sccp-b mainling lists on sourceforge. There
is active development in SVN but not in tarball releases.
http
having to add in anything extra to make everything work ok?, if not, is
there a version that someone may have carried forward of the skinny
driver that will work with 1.4?
Thank you,
Wayne.
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When I record calls they turn out really quiet. I can barely hear the
person on the other end. But when making the call, I can hear the
person on the other end just fine. I've played with txgain/rxgain and
that doesn't seem to help. TIA for help.
phones have no such difficulty
The only real console output is a mention of the peer being unable to
authenticate (which is odd since it's already registered)
This did not start occurring until our upgrade to 1.4.13 of Asterisk
Thanks for any insight you can provide,
Wayne
must admit - must
get round to having a looksee :)
Wayne.
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that - yes
things will be played in the correct order - but as each caller gets the
moh - it would 'carry on' at the position it was last at when the
/previous/ caller left moh - not from the start.
Cheery
Wayne.
David Gomillion wrote:
Hi All,
I need to have the same file played from
is 10sec long). It goes immediatey from receipt of the GET DATA
command to executing the next command, not even a return value.
Ha anyone seen this before? What can I do to get this working?
Thanks,
Wayne
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I was wondering where 3Com were getting all the new ideas from for their
phone system ;-p
Cats out of the bag now I guess :)
[EMAIL PROTECTED] wrote:
I will be out of the office starting 07/06/2007 and will not return until
07/09/2007.
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FindMe: 678.229.1809
http://www.sheltonjohns.com
- Original Message -
From: Wayne Jensen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 24, 2007 7:24:26 PM (GMT-0500) America/New_York
Subject
Every once in a while for no apparent reason, Asterisk has been dying
on me, dropping all calls in progress. There's nothing in the log
file or on the Asterisk console that indicates the reason. Some days
it doesn't happen at all. Other days it happens two or three times.
The problem began on
On 2/8/07, Wayne Jensen [EMAIL PROTECTED] wrote:
On 2/8/07, Don Pobanz [EMAIL PROTECTED] wrote:
Asterisk is getting red alarms on my T1, sometimes once or twice a
day, but today it happened 5 times. Even once is too many. Every
call in progress is dropped.
Red alarm means
On 2/8/07, Don Pobanz [EMAIL PROTECTED] wrote:
Asterisk is getting red alarms on my T1, sometimes once or twice a
day, but today it happened 5 times. Even once is too many. Every
call in progress is dropped.
Red alarm means that the hardware is not seeing the T1 signal coming in.
This most
Asterisk is getting red alarms on my T1, sometimes once or twice a
day, but today it happened 5 times. Even once is too many. Every
call in progress is dropped. Please help! What do I need to do?
What can I try? I've googled and searched this list and can't find
anything. Here's an example
On 2/7/07, Shane Spencer [EMAIL PROTECTED] wrote:
I had this happen to me because I was not configured properly and for
some reason the telco was automatically dropping me every so often. I
called the telco and they corrected me.
Shane
___
On 1/25/07, Wayne Jensen [EMAIL PROTECTED] wrote:
I'm having various issues that may or may not be related to each other (I'm
pretty sure they are). We've had this system for a year now (quad T1 card,
right now we have 1 T1 coming in, 2 going out to channel banks) and we've
had intermittent
' so I followed the instructions in README.udev
the error message went away, but now when I run ztcfg I just get 0
channels configured
Thanks!
Wayne
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yes
On 1/31/07, C F [EMAIL PROTECTED] wrote:
Is udev running?
On 1/31/07, Wayne Jensen [EMAIL PROTECTED] wrote:
I pulled a working TE405P from one box and put it in another box. I
compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no
lights on the card come on.
I do
On 1/31/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Jan 31, 2007 at 08:40:59PM -0700, Wayne Jensen wrote:
I pulled a working TE405P from one box and put it in another box. I
compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no
lights on the card come on.
I do
I'm having various issues that may or may not be related to each other (I'm
pretty sure they are). We've had this system for a year now (quad T1 card,
right now we have 1 T1 coming in, 2 going out to channel banks) and we've
had intermittent ghost calls--it appears that what is happening is a
at Astricon 2 years ago, but that could just be a
trick of my fading memory.
Digium and the community:
Thank you for Asterisk!
Also, just to clarify, although we work tightly with eBay, we are a separate
entity.
--
Wayne Walker
Operations Manager
UnWired Buyer, Inc.
http://www.unwiredbuyer.com
Wayne
.
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.
Believe me I'm quite mystified right now, but I'm worried if I look
into it too deeply it'll stop working for me, so I'm not asking too
many questions just yet :)
--Wayne
On Jul 20, 2006, at 7:43 AM, Mindaugas Kezys wrote:
How did you managed to et info who pressed the button for feature
an accessible either as a variable or through agi) which
party in a call is the party who actually invoked the feature?
Thanks for the help
Wayne
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heh. Just got permission to drop 1.2.10 on there and it seems to be
working the way i want it now.
--Wayne
On Jul 19, 2006, at 3:40 PM, Wayne P. HIll wrote:
I'm working on a server being implemented for a client right now
which, due to a long string of issues I won't go into, has decided
On Thursday 18 May 2006 03:35, Mark Coccimiglio wrote:
Otherwise the Diva server cards
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!
--
Cheers
Wayne
Hi all
Which command do I use to pull the mISDN number from an incoming call.
--
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Wayne
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Hi all
Does Digium make a quad BRI card? I can't see anything of the sort on their
page but I thought they might call it something else in the States.
Failing that, can anyone recommend a make/model that would handle 4 BRI ports?
--
Cheers
Wayne
...
--
Cheers
Wayne
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On Sunday 30 April 2006 10:27, Boris Bakchiev wrote:
Opened pseudo zap interface, measuring accuracy...
This may be a stupid question but how did you do this?
--
Cheers
Wayne
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£50). Like I say though - this was about 6-8 weeks or so ago since I
took delivery - I haven't checked to see if they are still selling them.
Wayne.
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Hiyall,
I don't suppose anyone has the elusive 'administrators' manual for these
things - I've got the users manual but would still like the full suit so
to speak.
Cheers
Wayne.
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dialled twice. I am using SCCP with Cisco
phones on asterisk 1.2.4.
Its as thought asterisk doesn't think anything is going on.
any thoughts?
Cheers.
Wayne.
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I say - I've never tried this - its what I
found out from the owners forums.
HTH.
Wayne.
Jonathan Attwood wrote:
I'm in conversation with Draytek's pre-sales dept..
Here's the most recent reply:
Hello,
We really don't know of anyone who has run an Asterisk server on
a Vigor2900
extension sits
and listens to the ringing.
does that make sense!
Wayne.
Matt wrote:
Ok.. here is another pet peeve i have.. that maybe someone can answer...
When I do a call file.
How can I make the call be transfered to the second party BEFORE the
first party picks up? In other words.. right now
yeah - that'll teach me for jumping in without understanding where your
coming from :)
/me crawls back under the rock...
Wayne.
Matt wrote:
Yeah.. except not for a busy-call-back type of application... or
really any application where you only want your phone to ring when the
line you
Forgive me if this is old news...
http://www.spectrum.ieee.org.nyud.net:8090/oct05/1846
--
Cheers
Wayne
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:43 DEBUG[18523] app_rxfax.c: Got hangup
Nov 24 11:12:43 DEBUG[18523] app_macro.c: Extension s, priority 3 returned
normally even though call was hung up
Nov 24 11:12:43 DEBUG[18523] pbx.c: Extension in_fax, priority 2 returned
normally even though call was hung up
--
Regards
Wayne Gemmell
..
As of my last posted log it was enabled and set to 15.
--
Regards
Wayne Gemmell
Work: +27 11 894 2530
Fax : +27 11 894 4081
Cell: +27 83 666 3325
Email : [EMAIL PROTECTED]
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Asterisk
On Thursday 24 November 2005 12:49, Kristof Hardy wrote:
Yes, make a 'default' to go directly to your fax-receive macro. (rxfax
witht the parameters)
At least you should hear a 'fax' answering.
Yes, I hear a fax answering, so at least I know its working.
--
Regards
Wayne Gemmell
Work
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=15.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
;busydetect=yes
;busycount=15
faxdetect=incoming
/zaptel.conf
--
Regards
Wayne Gemmell
On Monday 21 November 2005 17:12, Goran Donev wrote:
How do you install AMP? I downloaded it and tried to run make or install
and it doesn't work. Is there some trick to this?
The trick is to run the install script and read the documentation. Just not
in that order...
--
Cheers
Wayne
Well,
My votes goes to Allison!
Allison Smith wrote:
Asterisk Community Members:
What a day we've all had! OK -- deep breath -- relax, and let's take a
look at the situation calmly and rationally.
you know the rest
Love you all! You're all perfect lambs!
Allison
or something,
could be in the old handbook or hitchikers guide to asterisk as I havn't got
far enough into the new handbook to comment.
Based on your post, seems that you have an issue with codecs more than
creating an IAX trunk.
Thanks, yes I was disallowing all codecs, :(
--
Cheers
Wayne
on the
side I'm working from is as follows
Name/UsernameHost Mask Port Status
wayne165.165.164.87 (D) 255.255.255.255 4569
Unmonitored
and on the other side iax2 show users shows
Username SecretAuthen
- and I dont have skinny to put back! Can you even swap
these things back again?!
Wayne.
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it to work.
Any help gratefully received.
As a quicky - if you are using NTL - you don't need to patch - works
'out of the box'. NTL apparently use US (Bell?) standards for Caller ID.
I don't know about Telewest tho.
Wayne.
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into that gnomeeting CVS idea.
--
Regards
Wayne Gemmell
Tel Fax: (011) 894-4081
Cell : 072 836 4325
Email : [EMAIL PROTECTED]
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Hi all
Can anyone recommend a good soft phone that can compile on x86_64 (linux)
platform?
--
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Wayne Gemmell
Tel Fax: (011) 894-4081
Cell : 072 836 4325
Email : [EMAIL PROTECTED]
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processor runs on VXWorks but to cross the boundary of proprietary 3com
and rest of world - they jump onto windows.
Curiously
Wayne.
ps I don't know a great deal about the cisco system - its more hearsay
so please jump in on :)
Patrick wrote:
Reminds me of an Internet Call Diversion pilot
How about someplace central like South Africa?
--
Regards
Wayne Gemmell
Tel Fax: (011) 894-4081
Cell : 072 836 4325
Email : [EMAIL PROTECTED]
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on this list that at most
two could coincide in a box simultaneously without causing an interupt flood.
1) is my info okay so far?
2)What would be the best way for be to implement the other 22 lines? Is there
hardware I'm not aware of?
--
Regards
Wayne Gemmell
Tel Fax: (011) 894-4081
Cell
On Thursday 08 September 2005 16:26, Simone Cittadini wrote:
My boss is just asking me if it is possible to stuck 4* TE411P in a
Doesn't that equal 16 lines, not 480 lines? Or did I miss something?
--
Regards
Wayne Gemmell
Tel Fax: (011) 894-4081
Cell : 072 836 4325
Email : [EMAIL
cards in a single server is not safe. 1 per
server would be more acceptable.
This link might be helpful to you:
http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large
Thanks for the info, its exactly what I'm looking for. I knew things didn't
make sense.
Thanks
Wayne Gemmell
I think it would help if you sent an excerpt from your maillog.
Cheers
Wayne
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the card
wired in wrong and it didn't stop anything from working or being blown
so as a quick test ...
Wayne.
Giorgio Incantalupo wrote:
Hi,
we had 3 analog lines but our telco did't pass us the caller-id so
Asterisk tried to identify the caller-id but found bad data. Setting
usecallerid
Mahmoud Badran wrote:
Hello;
i am trying to make a dial plan that can handle any wrong extensions
dialled from the local sip phone for example so that if i dialled the
right extension it rings but if i dialled wrong or existing extension
it redirect him to the Main menu for example?
thanks
TELUX wrote:
does this version work? after the asterisk MIXING of files i have a
file of dead air.
Hmmm, in a twisted sort of way I'm glad to hear you're seeing this. I am
too and have been losing hair with it.
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useful app indeed! Do you plan to share that, sell
it, ??
Love to get more info or help..
Cheers,
Wayne
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Hiyall.
Just been testing my * to FWD connection again over IAX2
Well - I can get to 'tell me' and a UK freephone number so it would
appear its back up again. (ive not yet been lucky enough for anyone to
call me on it yet)
Thanks whoever :)
Wayne
I think you need to Answer the call after Wait ing for the CID.
Ronald Wiplinger wrote:
Calling from PSTN let extension 601 ring twice, hang up and starts
over again to ring twice, ...
If I pickup I do not hear on extension 601, and on the PSTN it is
still signaling to ring.
Can anybody
.'
Not heard anything back yet...
Wayne.
Jerry Glomph Black wrote:
This has been dead for days. What is FWD trying to do?
This kind of non-reliability makes the service worthless (or worse),
and is a disservice to VoIP in general. I thought Pulver's idea was
to get everybody to use VoIP
Carmi Weinzweig wrote:
On Nov 21, 2004, at 11:15 AM, Wayne Sheppard wrote:
Tracy R Reed wrote:
On Sat, Nov 20, 2004 at 06:55:56PM -0500, Noah Miller spake thusly:
This does seem to be a common request, but I haven't seen any great
Yes, it is. I am surprised * still can't do it.
I'm not surprised
Tracy R Reed wrote:
On Sat, Nov 20, 2004 at 06:55:56PM -0500, Noah Miller spake thusly:
This does seem to be a common request, but I haven't seen any great
Yes, it is. I am surprised * still can't do it.
I'm not surprised. Asterisk is a PBX, not a key system or a hybrid
system. The
Steven Critchfield wrote:
On Tue, 2004-11-16 at 09:53 -0600, Jay Milk wrote:
So, that's how my tax dollars are spent? Outrageous, and certainly
news-worthy. Good luck fighting off CNN and the like when this leaks
out.
It is covered under the No Child Left Behind program under continueing
TELUX wrote:
ive noticed that our 1.02 stable box's asterisk is taking 100% cpu load..
is there a problem with the stable version?
I've noticed this a few times, and it seems to depend on how * was started.
Try to start 'safe_asterisk' and see if it changes anything.
hiya,
I guess no help if you use BT - but I have got NTL (and you will need to
check that NTL in your area does CLID!) working with an un patched
version of * without any problems :)
Wayne.
James Botham wrote:
All,
Has anybody had any luck using the diffs for UK caller ID on the
latest CVS
the number you dialed is busy.
Thanks
Wayne.
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(if this can be tracked, not sure).
If you would like, I can send you some example reports that are used in
a typical call center, contact me directly if you would find that helpful.
Cheers,
Wayne
Ben Merrills wrote:
I've been doing some work on a queue log analyser for a while now,
getting the basics
I would suggest the following, but these are pretty standard in a call
center -
logged in/out
ready/not ready (not ready reason code support, i.e. call wrap, break,
training, etc.)
on inbound call
on outbound call
recording call flag
stats at agent level -
total inbound calls
total outbound
to solve that problem ?
Let me know if you need copy of zapata.conf, zaptel.conf, voicemail.conf
and extensions.conf
Thanks in advance and a very thanks for the developper for that incredebly
very nice software running on Linux :)
Wayne
*CLI -- Starting simple switch on 'Zap/1-1'
Sep 22 16:29:33
is the lsof output when * is running:
lsof | grep dsp
asterisk 14502 root 16u CHR 14,3 2489 /dev/dsp0
Very thanks in advance.
--
WayComm
Wayne Veilleux ing., GCIA, CISSP
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Thanks Marconi, that solve my problem.
Bye.
Wayne
On Tue, 14 Sep 2004 06:47:53 -0400 (EDT), Wayne Veilleux
[EMAIL PROTECTED] wrote:
Hi,
I'm new to *. I just installed my X101P card with * from the source on
Mandrake 10.0 and I test it. Everything seems to work fine. When I call
at my
and 104 to which they then have to go into their mailbox and delete the
message?
(hope that makes sense)
Wayne
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=101,102,103 in zapata.conf, sip.conf, etc...
-Seth
Hy - nice one - worked for me - Cheers Chap!
Wayne.
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installed.
Thanks
Wayne.
Stephen Hon wrote:
Hi..
Has anybody been experiencing any problems with transfers using # to
transfer after taking a call off of hold?
Transfers using the # and music on hold work fine by themselves.
However, when we place somebody on hold we can no longer use
that when you dial '*8#' - asterisk is only getting a
'*8' and not knowing what to do with it.
Dunno if you can change a cisco to not use # to 'send' - too new to all
this at the mo - this is just what I've observed with playing at home :)
Wayne.
Nicolas Gudino wrote:
Hello,
On Wed, 2004-08-04
- hope this helps :)
Wayne!
[EMAIL PROTECTED] wrote:
Hi Sean
Both phones are set for context=sip in the sip.conf file.
As I say the phones will both call out OK (I can dial the 500 test number and
successfully connect to the remote PBX through my firewall). It's just that when I'm
trying to call
Hiya,.
What cisco phone / firmware are you using? - Ive got a 7960 with SIP and
the only 'soft' button that comes up when you get an incomming call is
to 'answer', havnt got an option for 'send to VM'
Thanks
Wayne.
Brian wrote:
; Below assumes you are using the same number for Voicemail boxes
Hiya,.
What cisco phone / firmware are you using? - Ive got a 7960 with SIP and
the only 'soft' button that comes up when you get an incomming call is
to 'answer', havnt got an option for 'send to VM'
Thanks
Wayne.
Brian wrote:
; Below assumes you are using the same number for Voicemail boxes
the branch office.
any thoughts.
Wayne.
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Can anyone point me in the direction of a Asterisk
developer in New Zealand that we could contact???
Many thanks
Wayne Methorst
New Zealand
[EMAIL PROTECTED]
We've got one of the Budgetone phones here, and we can call from any SIP
phone, or an outside line TO this phone and the conversation sounds great for
bothways, not a bad delay, no echo problem, etc. But when we pick up the
Budgetone and dial an outside line or another SIP phone the person on
on the
network. As all our PC's on the network use Microsoft OS is there a free
gatekeeper software for microsoft operating system?
Does Asterisk have a built in gatekeeper?
Many thanks
Wayne
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