I have a similar problem in Australia and I think it has to do with
chan_zap.c
Currently Digium are investigating it for me as it is in association
with one of their TDM400P cards.
Gonzalo Servat wrote:
Hi there.
I'm having a strange issue with the distinctive ring detection in
Asterisk (I
Compile CVS HEAD and it's all built in.
Andy Kuo wrote:
Hi,
I've been trying to get fax going for the last few days.
I can receive fax with Spandsp 0.0.2pre19 on Asterisk 1.0.9 with rxfax,
but when I tried sending the received fax file to a fax machine, I
either get line error or just a
Has anyone got distinctive ring detection working for PSTN lines in
Australia.
I am using the latest CVS and have got zapata.conf set up thus: but it
appears that the chan_zap modules is not going anywhere near that piece
of code and all it returns is the default 0,0,0
[channels]
context =
I think one of the most important and flexible features of * is the
ability to restructure the dial plan by using the n and s priorities.
I cannot, for the life of me, see why they only exist in the CVS strand
and not in the release strand; even the CVS of the release strand
doesn't have
checkout zaptel libpri asterisk
So, when does a feature cease to be a feature, and what does it become?
Kevin P. Fleming wrote:
Howard Lowndes wrote:
I cannot, for the life of me, see why they only exist in the CVS
strand and not in the release strand; even the CVS of the release
strand
version. I have certainly found them to be both
essential and stable.
Kevin P. Fleming wrote:
Howard Lowndes wrote:
So, when does a feature cease to be a feature, and what does it
become?
I don't understand the question...
___
--Bandwidth
Great :)
Kevin P. Fleming wrote:
Howard Lowndes wrote:
This implies that n s priorities are features since they are
not in 1.0.x. So, when do they cease to be features and become a
standard part of the released version. I have certainly found them to
be both essential and stable
I'm banging my head against a brick wall trying to get CallerID
recognised in Australia.
I have CLID presentation enabled and I know that it works. I also have
distinctive ring tones enabled in zapata.conf
Around about line 5924 in channels/chan_zap.c is where the caller ID and
distinctive
It's also catering for the fact that ${FROM_DID} might be a string with
embedded spaces, and it's assuming, probably not unreasonably, that
${CALLERIDNUM} doesn't have embedded spaces.
David Tillman wrote:
In my (inherited) extensions.conf I have some lines of the format:
exten =
This is not a newbie question, and my problem may be related to
Australia only or may be wider based.
I have a PSTN line that has Caller ID presentation enabled.
It used to work fine until recently, in as much as I could identify
inbound CLID.
There is/was a patch to * that suggested that
Is there any chance anyone could discuss my post under Can someone
please explain caller line identification. I live in Albury so I have
no chance of getting to ML tonight.
jurgen wrote:
Just a quick reminder - this is happening *TONIGHT*. Hope to see all
local Asteriskers come out (except
Koolstart - see attached
Rudolf Ladyzhenskii wrote:
Hi, all
I have installed TDM400 with 1 FXS and 1 FXP ports.
Now I am goig through documentation on how to configure it.
It mentions 3 protocols: Loopstart, Groundstart and Koolstart. Which one
do I use?
Can someone send me sample
This is not a newbie question...
I have CLID presentation enabled on my PSTN service for some months now
and it has worked fine until the other day - I could discover the
CALLERID and hence could divert the telemarketers to voicemail.
I then did a routine update of Linux and recomplied * -
Grandstream supports DNS SRV
Justin Richards wrote:
Have you found any information yet about this? I am looking for good
and affordable phones that can use DNS myself, but not for failover,
simply for ease of use by some non-computer savvy family members. So
far, I am afraid I'm going to be
This is an Australian situation.
I have a PSTN connection that has CLID presentation enabled and has two
numbers assigned to it, the primary number with the standard ring
cadence: 400,200,400,2000 and the secondary number with the alternative
cadence: 200,400,200,400,200,1600
CLID
On Fri, 2005-03-18 at 08:34, Asterisk wrote:
John Goerzen wrote:
Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:
exten = s,1,SetVar(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten =
as asked.
--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft.
On Thu, 2005-03-10 at 08:05, Martijn van Oosterhout wrote:
Hi,
I see applications for signalling busy, congested, ringing, progress
etc, which I understand can be provided either in or out of band. But
all I want to do is generate a dialtone. This obviously can only be
done in band.
On Wed, 2005-03-09 at 05:29, kurt x wrote:
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding is
this venture. When I dial and access voicemail with an ani of 3000
the gotoif statement does not push the call to s|6. Its goes through
each line( 1,2,3,4,5,6,7) .
On Tue, 2005-03-08 at 09:14, Anton Krall wrote:
Guys.
Im trying to implement some kind of call forward or DND, I checked the wiki
and there are some examples of call forwards but I was wondering if anybody
has implemented one that will let you forward calls to SIP, IX or ZAP
channels alike?
-ext:s-int)
Sounds good?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes
Sent: Lunes, 07 de Marzo de 2005 04:43 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Forward or DND
On Tue
On Tue, 2005-03-08 at 11:43, Anton Krall wrote:
Wow, too professional for me hahaha can you explain to me the last part of
the goto?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes
Sent: Lunes, 07 de Marzo de 2005 06:22 p.m
On Tue, 2005-03-08 at 14:57, Brian Nehring wrote:
I actually got X-Lite talking to the server, finally. I didn't have to
change any of my Asterisk servers... I just kept fooling around with
X-Lite and watching the diagnostics log and it finally worked. I can't
really say what fixed it, I don't
On Tue, 2005-03-08 at 16:56, Brian Nehring wrote:
Xlite for OS X actually.
bummer, I've been wanting to get it running under Linux.
On Tue, 08 Mar 2005 15:00:24 +1100, Howard Lowndes [EMAIL PROTECTED] wrote:
On Tue, 2005-03-08 at 14:57, Brian Nehring wrote:
I actually got X-Lite
On Mon, 2005-03-07 at 09:02, David Newman wrote:
On Sun, 6 Mar 2005, Marty Mastera wrote:
The full text of the bug you reference above indicates that pstnVMgain
was (or is) part of an ongoing feature request/bug report and has not
been implemented for use at this time (and may never be).
On Sat, 2005-03-05 at 14:10, Anton Krall wrote:
I think I have something misconfigured regarding voicemails. They work
great, I have this setup:
Sip.conf
[ext1]
Context=phones
Mailbox=201
Voicemail.conf
[home]
201,password,name,[EMAIL PROTECTED]
Voicemail delivery and all
On Thu, 2005-03-03 at 06:32, Randy Johnson wrote:
Is there a way to send a voicemail to two different email addresses when
a caller leaves a message?
Does address1, address2 work or does it get confused about the ,?
Thanks a bunch!
Randy
___
= on
channel = 1
channel = 2
channel = 3
channel = 4
Any ideas?
- Original Message -
From: James Andrewartha [EMAIL PROTECTED]
Sent: Tuesday, January 04, 2005 11:50 PM
Subject: [Asterisk-Users] CallerID in Australia Analogue PSTN PhoneSystem
Howard Lowndes wrote
Does anyone know why the n priority in the dial plan is not recognised
in 1.0.6 It seems strange to me that it should be so.
--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just want a system that works,
Has anyone had problems with Call Waiting signals causing Zap channel or
bridging hangups in AU.
I was on a call the other day (Zap channel to PSTN) and the call
suddenly hung up on my side. I dialled the calling party and got the
call again, it seems that the bridge had dropped and that the
On Mon, 2005-02-28 at 20:38, Azhar Chowdhury wrote:
I would like connect two offices where one office have 4 PSTN Analog lines
and another office without any PSTN. Both the offices will have two separate
Asterisk server with TDM400P cards (4 ports FXS FXO).
My questions is that how to
Primary * box detects DD0S - runs:
asterisk -rx database put PANIC DDOS YES
and have your dialplan look for that database family/key being set to
determine which path it takes.
When the primary * box detects that the DD0S is over - runs:
asterisk -rx database del PANIC DDOS
On Tue,
On Tue, 2005-03-01 at 07:11, Colin Anderson wrote:
How about a combination of GotoIF, and app_dbodbc (or app_db):
exten = 700,1,playback(ddos-on)
exten = 700,2,DBput(DDOS/yes)
exten = 701,1,playback(ddos-off)
exten = 701,2,DBdel(DDOS/yes)
[mymainaa]
exten = s,1,DBGET(TRUE=DDOS/yes)
On Tue, 2005-03-01 at 08:03, Kristian Kielhofner wrote:
Tony Mountifield wrote:
I've just set up a new box with FC1+updates and the latest Stable
Asterisk from CVS.
Why are you using FC1 when FC3 is out? Better yet, why are you using
FCx at all?
Why not? What are you, some sort
On Mon, 2005-02-28 at 05:58, Riphagen, Ferdy wrote:
Hello All,
I have a macro and want to jump to another macro if a conditition is true or
false.
Asterisk is jumping to the next macro, but then the {ARG1} variable is not
working anymore.
Try SetVar(SAVEARG=${ARG1}) in one macro then
On Wed, 2005-02-23 at 01:52, Mark Eissler wrote:
On Feb 21, 2005, at 7:35 PM, Rudolf Ladyzhenskii wrote:
Hi, all
I am doing prrof of concept system. I will have two IP phones
connected to Asterisk box. Box itself will have 1 PSTN conenction and
one analog phone conenction. A basic
On Tue, 2005-02-22 at 11:41, John Novack ( Mozilla - portable ) wrote:
Rudolf Ladyzhenskii wrote:
Hi, all
I am doing prrof of concept system. I will have two IP phones connected to
Asterisk box. Box itself will have 1 PSTN conenction and one analog phone
conenction. A basic minimal
On Mon, 2005-02-21 at 15:50, dkwok wrote:
How to announce the DNID to the called party who picks up the phone and
say the correct greeting?
I suppose it has to say to the called party before the call is bridged.
So it has to do something before the dial command transfer the call.
Any
On Thu, 2005-02-17 at 15:24, dean collins wrote:
http://www.srh.noaa.gov/fwd/productviewnation.php?pil=OKXZFPOKXversion=0
can anyone suggest how I could set up [EMAIL PROTECTED] to read out
allowed the following text when I dial extension 850?
815 PM EST WED FEB 16 2005
On Thu, 2005-02-17 at 15:51, [EMAIL PROTECTED] wrote:
I've installed a TDM400. Having a go with AMP.
I would like incoming calls to be put throuhg to an extension (at my desk)
and a mobile (cell phone) at the same time. Whichever picks up, gets the
call..
This should be possible with AMP
On Wed, 2005-02-16 at 09:33, Stefan Gofferje wrote:
Rod Bacon schrieb:
I know this is casting a wide net, but If you were charged with building
a large, public VOIP network with multiple PSTN gateways, the capacity
to carry a lot of traffic and bill clients accurately, what pieces
On Wed, 2005-02-16 at 11:05, Greg Oliver wrote:
OK - I can successfully make calls from SIp phone through an asterisk
323 channel to a Cisco Call Manager and out a MGCP controlled gateway.
The problem is that if the call is not answered within ~5 seconds, *
gives the message no one is
On Wed, 2005-02-16 at 13:07, Shaun Ewing wrote:
On Wed, 16 Feb 2005 12:20:00 +1100, Paul Hales [EMAIL PROTECTED] wrote:
Regarding your quote about Polycom - I'm not sure what you mean by 'Polycom
won't sell...'
We have over 100 polycom's out and about, all hooked into our 3 Asterisk
On Wed, 2005-02-16 at 13:14, Shaun Ewing wrote:
On Wed, 16 Feb 2005 09:23:21 +0800, Stuart Elvish [EMAIL PROTECTED] wrote:
Definitely agree - don't even try using the Grandstream for a
receptionist (among other things the phone probably won't hold out
physically for more than a few weeks if
On Tue, 2005-02-15 at 15:55, Brian Dingman wrote:
Wow. I posted that a long time ago. Thanks. Festival doesn't seem very
stable to me though.
Works fine for me, but I think the non-US accents need some work.
On Tue, 15 Feb 2005 15:14:47 +1100, Rod Bacon
[EMAIL PROTECTED] wrote:
SIOD
On Tue, 2005-02-15 at 17:13, Rudolf Ladyzhenskii wrote:
Hi, all
I am in Australia and I have to setup Asterisk in few offices. There will be
IP phones in each office and I must be able to call between offices.
I use the HOP 1002 from IP Trading in Sydney - I think they call it the
Vision
On Tue, 2005-02-15 at 17:26, Paul Hales wrote:
The Asterisk meeting in Melbourne Thursday night would be a good place to
discuss this!
Not if:
1. You don't know about it
2. You're not Melb based.
Regards,
regards,
PaulH
-Original Message-
From: [EMAIL PROTECTED]
On Tue, 2005-02-15 at 17:43, Adam Goryachev wrote:
On Tue, 2005-02-15 at 17:13 +1100, Rudolf Ladyzhenskii wrote:
Hi, all
I am in Australia and I have to setup Asterisk in few offices. There will
be IP phones in each office and I must be able to call between offices.
I need actual
On Tue, 2005-02-15 at 18:05, Adam Goryachev wrote:
On Tue, 2005-02-15 at 17:54 +1100, Howard Lowndes wrote:
On Tue, 2005-02-15 at 17:43, Adam Goryachev wrote:
Personally, I quite like the polycom phones such as the IP300 and IP600
I've never really bothered with the IP500
On Mon, 2005-02-14 at 10:10, Gary wrote:
On Mon, 14 Feb 2005 09:53:36 +1100, PHP Mechanic wrote:
http://www.broadbandphone.com.au/global/pnp.htm
They look like they are all PA1688 based.
The black one is a dead copy of the one sitting on my desk, made by
Hirakawa Electronics according to
On Mon, 2005-02-14 at 13:52, Craig wrote:
Message: 1
Date: Mon, 14 Feb 2005 09:53:36 +1100
From: PHP Mechanic [EMAIL PROTECTED]
Subject: [Asterisk-Users] Who makes these phones?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
In AU we have a number of different dialtones defined for various
purposes.
From indications.conf:
au ringcadance 400,200,400,2000
au dial413+438
au busy425/375,0/375
au ring413+438/400,0/200,413+438/400,0/2000
au congestion
On Tue, 2005-02-08 at 09:49, Brian Dingman wrote:
I am having the same problems. No matter what I try, * won't detect
faxes. I have faxdetect=both in zaptel.conf and my extensions.conf
looks like this:
[fromPSTN]
exten = s,1,Answer
exten = s,2,DigitTimeout(2)
exten =
On Tue, 2005-02-08 at 09:49, Brian Dingman wrote:
I am having the same problems. No matter what I try, * won't detect
faxes. I have faxdetect=both in zaptel.conf and my extensions.conf
looks like this:
[fromPSTN]
exten = s,1,Answer
exten = s,2,DigitTimeout(2)
exten =
On Sat, 2005-02-05 at 08:28, Eric Bishop wrote:
Hi All,
I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and
out of Asterisk. I am using PSTN Ring Thru Line 1 (on the PSTN
Line tab) so Asterisk answers the call rather than the SPA-3000. It
is all working perfectly except I
On Sat, 2005-02-05 at 12:33, Steven P. Donegan wrote:
I have done that extensively (H.323 and SIP over IPSEC tunnels) I was
more interested in the possibilities of 'native' support of some kind.
But thank you very much for the response.
Isn't there a fairly significant overhead with this,
On Thu, 2005-02-03 at 07:07, Martin Roy wrote:
OK I have 12 phone lines connected to 3 digium TDM04B cards on the same
server. I must do the following thing :
The first 10 lines will be use by one company and the 2 left by another
one. For outgoing calls it's quite easy I just create 2
Surely there has to be one soft phone that works under Linux.
I've tried:
kphone - it sometimes complains about the need to release the sound
device
linphone - lowww
iaxcomm - needs some strange widgets
various others - either only supplied as binaries, or just plain
On Wed, 2005-02-02 at 07:41, Michael Van Donselaar wrote:
On Wed, 02 Feb 2005 07:12:54 +1100, Howard Lowndes [EMAIL PROTECTED] wrote:
Surely there has to be one soft phone that works under Linux.
I've tried:
kphone - it sometimes complains about the need to release the sound
device
= yes
callerid = asreceived
;usedistinctiveringdetection = yes
useincomingcalleridonzaptransfer = yes
faxdetect = both
group = 1
channel = 4
Best,
...jurgen
On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote:
Is anyone having/had a problem with a TDM400P card
On Tue, 2005-02-01 at 08:12, Eric Bishop wrote:
Has anyone implemented hot seating in any neat way? This where
people can log in to any phone in the company and have their
calls/voicemail come to that particular handset.
I think this usually called follow me and is a variation on call
/Perth, Howard Lowndes wrote:
Is anyone having/had a problem with a TDM400P card hanging up on STD
outbound calls as soon as the called party answers.
I'm guessing that * is responding to the STD pips in some way.
--
Howard.
LANNet Computing Associates;
Your Linux people http
in zaptel.conf.
Yes I have that set and similar in indications.conf.
This should tell asterisk to look for Australian tones rather than the
US ones which I assume it does by default.
Hope this helps.
Kind Regards
Stuart
On Tuesday, Feb 1, 2005, at 10:46 Australia/Perth, Howard Lowndes
Is anyone having/had a problem with a TDM400P card hanging up on STD
outbound calls as soon as the called party answers.
I'm guessing that * is responding to the STD pips in some way.
--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
Of Howard Lowndes
Sent: Friday, 28 January 2005 17:30
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Caller ID in AU
Is anyone in AU successfully getting Caller ID from the analogue PSTN
service?
If so, what settings?
--
Howard.
LANNet Computing Associates;
Your Linux
On Fri, 2005-01-28 at 19:21, PHP Mechanic wrote:
Is anyone in AU successfully getting Caller ID from the analogue PSTN
service?
If so, what settings?
--
Howard.
http://www.voip-info.org/tiki-index.php?page=Asterisk+and+Australian+Caller+ID
Done that one already.
On Sat, 2005-01-29 at 05:50, Manjit Riat wrote:
Just installed festival from source and the voice is very jittery and
I get this a lot in the asterisk CLI (at least once on every call)
NOTICE[3236]: rtp.c:430 ast_rtp_read: RTP: Received packet with bad
UDP checksum
I get that also, but
On Fri, 2005-01-28 at 17:12, [EMAIL PROTECTED] wrote:
Folks,
I'd like to change the value of ${CALLERIDNAME} for incoming PSTN
calls from certain numbers, but haven't found a way that works. The goal is
to provide more informative names on my phones' caller ID displays--e.g., I
would
Is anyone in AU successfully getting Caller ID from the analogue PSTN
service?
If so, what settings?
--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just want a system that works, you choose Linux;
when you
On Thu, 2005-01-27 at 03:34, Michael Welter wrote:
Since we're chatting about tftp servers...
Let's say I have a new customer with Cisco 79xx phones, and he desires
to SIP register on my Asterisk system. I would have to provide the
SIPmac.cnf and SIPDefault.cnf files on my tftp server for
Is it possible to run the Festival command in the same manner as the
Background command so that it can be interrupted by caller key presses?
--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just want a system
The Dial command can be made to make an announcement to the called party
before channel is bridged.
Is it possible to make that announcement a Festival command in some way.
--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
On Mon, 2005-01-24 at 14:45, Gary wrote:
On Mon, 24 Jan 2005 14:57:06 +1300, Matt Riddell wrote:
Howard Lowndes wrote:
Is it possible to get the Festival command to read the text from a
system file rather than having it input as a text string?
Is this a case of having to use AGI
On Mon, 2005-01-24 at 21:37, Emanuele Venditti wrote:
Does anybody what the regional settings are to use an x100p (clone)
card
with Asterisk in Australia?
I got mine installed and recognised by * but I get no sound and
terrible hangup detection.
Basically after each test call to the
spell telco cartel?
Andrew
On 25/01/2005, at 2:25 AM, Howard Lowndes wrote:
On Mon, 2005-01-24 at 21:37, Emanuele Venditti wrote:
Does anybody what the regional settings are to use an x100p (clone)
card
with Asterisk in Australia?
I got mine installed and recognised by * but I get
On Tue, 2005-01-25 at 11:13, Ronan Mullally wrote:
I've got an old Dell Optiplex (Pentium-II, 1998 Vintage) which is
successfully running an X100P card. I'm hoping to upgrade to a TDM400.
Has anybody tried running these cards in old Optiplex machines? I'm not
particularly worried about
Is it possible to get the Festival command to read the text from a
system file rather than having it input as a text string?
I suppose I could put the text string into an Asterisk variable and
reference that in the Festival command, but then, how do I get the
contents of the file into the
17:02:44 +1100, Howard Lowndes [EMAIL PROTECTED] wrote:
I have a zap line on a X101P which will occasionally just hang up the
call for no apparent reason. Is there any good way of trying to
diagnose what might be causing this? Monitoring the asterisk output in
verbose mode does not provide
On Fri, 2005-01-21 at 14:32, [EMAIL PROTECTED] wrote:
Hi asterisk users!
Heres my issue, Ive deleted the s extension cause I dont want
any
action to be taken on incoming calls as my pbx is for home use, but I
would like to ring all my VoIP extensions at the same time the PSTN
line
I have a zap line on a X101P which will occasionally just hang up the
call for no apparent reason. Is there any good way of trying to
diagnose what might be causing this? Monitoring the asterisk output in
verbose mode does not provide any indications.
--
Howard.
LANNet Computing Associates;
I've just replaced a X101P card with a brand new TDM400P card
(specifically TDM421B).
I do have the molex plug attached.
kudzu removed the config for the X101P OK, but didn't find the TDM400P
lspci does not show the card
?? Bung card ?? How susceptible are these cards to XRays, as it has
been
On Tue, 2005-01-18 at 07:44, kurt x wrote:
I am trying to use the Directory() but am having difficulty using it.
According to Wiki page that I found you need to pass it
your voicemail.conf context. My vm-context is [local]. So when
I setup the cmd (Directory(local)) I can search on the
Will Wait(n) still listen for DTMF input from the caller after there has
been a Background(some-message) prompt, or do I need to use
Background(silence/n) to still listen for DTMF?
--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
On Tue, 2005-01-18 at 13:18, Eric Wieling wrote:
Howard Lowndes wrote:
Will Wait(n) still listen for DTMF input from the caller after there has
been a Background(some-message) prompt, or do I need to use
Background(silence/n) to still listen for DTMF?
The WaitExten and Read
Have a close listen to digits/h-15 and digits/h-16.
To my ears the latter could be mistaken for the former ... or perhaps I
am more deaf than I think.
--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just
Can anyone _recommend_ a downloadable OSS softphone that _works_ under
Linux and is compatible with Asterisk.
So far I have tried kphone and linphone and had problems with both, and
I am still waiting to hear back from the X-Lite beta folks.
--
Howard.
LANNet Computing Associates;
Your Linux
On Sat, 2005-01-15 at 07:09, Adam Fineberg wrote:
Howard Lowndes wrote:
Can anyone _recommend_ a downloadable OSS softphone that _works_ under
Linux and is compatible with Asterisk.
So far I have tried kphone and linphone and had problems with both, and
I am still waiting to hear back
On Sat, 2005-01-15 at 12:27, Michael Van Donselaar wrote:
iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol.
It is distributed as part of Steve Kann's iaxclient library.
I've just posted new Windows, Linux and Mac OSX binaries to sourceforge.
The Windows binary
On Sat, 2005-01-15 at 15:03, Philippe Daoust wrote:
Hello list,
I want to listen to voicemails on my * box from a phone that is not
local to my pbx. I.e., from my cellphone or my PSTN work line etc. I'm
aware that I can forward VM to email or use a web interface but that is
not always
I have echo training set on in my zapata.conf file for a X101P card:
echocancel = yes
echocancelwhenbridged = yes
echotraining = yes
Now, I know that echo cancellation is a black art, but I am finding that
at the beginning of a call bridged between a SIP channel and a Zap
channel the voice
On Fri, 2005-01-14 at 06:20, Keith LeClaire Jr wrote:
I'm trying to patch the current asterisk CVS with spandsp-0.0.1k.tar.gz.
Everything compiles fine but when I go to patch the asterisk/apps/Makefile
it fails:
asterisk:/usr/src/spandsp2# patch Makefile.patch
can't find file to patch at
On Fri, 2005-01-14 at 14:14, Mike Boger Jr wrote:
Hi,
Here's the deal. When someone leaves me a voicemail message I want
Asterisk to call me on my cellphone by dialing my cellphone number and
tell me I have a message. Is this possible? Can anyone cite examples?
Most commercial voicemail
On Fri, 2005-01-14 at 15:09, Andrew McRory wrote:
I have uploaded kphone and asterisk CVS stable. These packages are built
for Fedora Core 1 and this asterisk release should fix the non-root
permissions problem I worte about...
ftp://ftp.linuxsys.com/pub/releases/FC1/
I have just
On Fri, 2005-01-14 at 15:09, Andrew McRory wrote:
I have uploaded kphone and asterisk CVS stable. These packages are built
for Fedora Core 1 and this asterisk release should fix the non-root
permissions problem I worte about...
ftp://ftp.linuxsys.com/pub/releases/FC1/
OK, there are a
On Thu, 2005-01-13 at 10:40, John Dunham wrote:
Just checking if anyone has experence with Integrated Networks IN1002 phone.
You might like to try aredfox.com and see if there is anything there
that might suit. I have HOP1002 phones and I am using the 1002 as a
clue here.
We just got 100 of
On Thu, 2005-01-13 at 11:11, Adam Goryachev wrote:
On Thu, 2005-01-13 at 10:20 +1100, Howard Lowndes wrote:
I have a situation where I need to know which Zap channel an incoming
call is on, so that the call can be answered appropriately when a SIP
phone displays the channel. These Zap
On Thu, 2005-01-13 at 12:38, Adam Goryachev wrote:
On Thu, 2005-01-13 at 11:24 +1100, Howard Lowndes wrote:
On Thu, 2005-01-13 at 11:11, Adam Goryachev wrote:
On Thu, 2005-01-13 at 10:20 +1100, Howard Lowndes wrote:
I have a situation where I need to know which Zap channel an incoming
On Wed, 2005-01-12 at 11:01, Matthew Boehm wrote:
what is g723? ive never seen that before...
It's a codec. and it look like you have some form of codec translation
problem.
-- Executing Answer(Zap/1-1, ) in new stack
-- Accepting call from '2819870065' to '2815692780' on channel 0/1, span
On Wed, 2005-01-12 at 12:40, Matt Riddell wrote:
Ferguson, Michael wrote:
G'Day All,
rpm -q kernel-source returns Package kernel-source is not installed
Where can I find it and install it. Asterisk evidently needs it for a
successful install.
You can do:
yum install kernel-source
On Tue, 2005-01-11 at 00:29, Rich Adamson wrote:
I am new to asterisk but learn a lot about it to this mailing list and
wiki currently i am facing problem about sip phone i have PA 1688
chipset ip-phone and i have iptel.org sip account i registered locally
and through iptel.org comfortably
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