On Thu, 2005-07-21 at 13:00 +0900, Vic wrote:
Dear all,
1. Person will call our phone number
2. He will be asked to press 1 for Office 1 map, 2 for Office 2 map
and 3 for Office 3 map.
3. User presses 1.
4. User is asked to enter his phone number.
5. User enters his phone number
On Wed, 20 Jul 2005, Ed Greenberg wrote:
I'd like to write a snippet of dialtone that plays dialtone and collects a
specific number of digits into a variable.
Sort of like READ but with a generated dialtone.
Naturally, I want the dialtone to stop playing after the first digit.
I can't
On Thu, Jul 21, 2005 at 09:59:05AM +0800, Ronald_Wiplinger wrote:
asterisk_on_oelf wrote:
Hi,
I have a soekris 4801 since some days. I use it with a FritzCard-USB
and an
internal HFC-Card (NT Mode). Everything is working, but I still havn't
had time
for performance test. Only thing I
On Thu, Jul 21, 2005 at 01:31:59AM +0800, chris wrote:
hi kevin,
i tried removing the enitre asterisk directory and upadatesd my cvs folder.
and try to run make.. i'm getting
make_version_h : cannot execute error
Maybe you misread the error?
Maybe this is an error from this script that
I have all my GXP-2000's set to dynamic with no problems. You need to make
sure they have the latest firmware, as this fixed a few issues and improves the
overall usage of the phone. Hopefully they will make the useless LED's work so
we can line monitor etc...
Regards
Lee
-Original
On Wed, Jul 20, 2005 at 01:41:52PM -0400, O'Neill,Davin S. wrote:
I previously had Asterisk 1.0.7 running on a Linux 2.4.x kernel with
ztdummy. I was able to do things like meetme and music on hold. I
recently installed Asterisk 1.0.9 on a different machine with a Linux
2.6.x kernel running
David Hajek wrote:
Yes, I tried signalling = bri_cpe_ptmp.
When I put the card into older system and use same cables, same ISDN
units, same Asterisk configs (but older bristuff!) it works fine. When I
put the card into Dell, I got the CRC errors as I wrote before. Maybe
someone from Junghanns
On Wed, Jul 20, 2005 at 02:13:57PM -0400, J.Raborg wrote:
Folks:
does anybody have an idea? how to capture the DTMF digits to a file, after
an extn asnwer? then POST it to a url?
Off the top of my head:
Read(DIGITSVAR)
System(echo ${DIGITSVAR} /path/to/file)
Curl(URL)
;or:
Hi to all once again
Thanks for you help. I always get my problem solved from here.
What i did.
export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
cvs login
cvs checkout -r v1-0 zaptel libpri asterisk
and then compiled each of them
then on running
/etc/init.d/zaptel start
Both of
Ronald_Wiplinger wrote:
asterisk_on_oelf wrote:
Hi,
I have a soekris 4801 since some days. I use it with a FritzCard-USB
and an
internal HFC-Card (NT Mode). Everything is working, but I still havn't
had time
for performance test. Only thing I tested, was two ISDN channels via
FritzCard
in
Hello,
I have the following setup:
sip phones -SER - asterisk - voip provider1
- voip provider2
i got a toll-free DID from voipprovider1 to allow people from outside
to call into asterisk, get authenticated, and use voipprovider2 to
call out (kind
hi Tzafrir,
i was able to run make by removing ^M at the end of each line of each
script, i also checked all script file on the /asterisk folder and execute
dos2unix command on all script files, however when i run make i encountered
another problem.
gcc -pipe -Wall -Wstrict-prototypes
I think that's mostly right, but it should also be a native
xfer function working the same way regarding of the user agent, some
sort of common ground we can trust for installation with mixed
devices.
By the way: anyone got experience in attended trasfer with snom ? :)
Alessio Focardi
PF
On Thu, Jun 30, 2005 at 09:45:56AM -0600, Joseph wrote:
I think this is a weak point in asterisk.
It doesn't even have a means of email notification if IAX or SIP
registration fails.
But it sends an error to the a log (configurable to some extent in
logger.conf).
tail -f /that/log/file | grep
Is there a possibility to send text based messages from/to a sip phone
(text display) or to a video phone or from/to a messenger?
bye
Ronald
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I am trying to find out what is the best board for us.
I want to build an asterisk based PBX with one digium TDM422 card.
A USB wireless adapter should make the entire system with:
* Asterisk at home (Is @home or regular better?)
I have no experiences with @home. I use a debian. Debian sarge
I know that this subject has been treated in the past!
As a matter of fact reading some old messages about compiling zaptel I
made a couple of tests after the first compiling failure to understand
why I can't compile on a specific machine, but I do not know how to
handle the results.
The machine
Ronald_Wiplinger wrote:
Is there a possibility to send text based messages from/to a sip phone
(text display) or to a video phone or from/to a messenger?
Yes, there is in SIP if the SIP user agents support it. But no, Asterisk
will not forward the SIP messages between the SIP user agents.
Rob Engstrom [EMAIL PROTECTED] wrote:
(Article auto-converted from unnecessary HTML to nice plain text.)
We've just setup our [EMAIL PROTECTED] server, with our quad port card.
Everything
works well so far.
One thing I notice is that when I leave the handset on the hook and dial
a #,
On Thu, 2005-07-21 at 08:14 +1000, Eric Bishop wrote:
Yes, I have some advice. Use Fedora Core 2. I have battaled for almost
a year to get fcpci and udev-based distributions working with very
limited success.
On 7/21/05, Adrià Vidal [EMAIL PROTECTED] wrote:
Someone have info about
2 to 4 channels to start with.
Allan.
--- hakem voip [EMAIL PROTECTED] wrote:
How many channels do you need per gateway ?
I might have slution for you voip2gsm
Regards
On 7/20/05, Allan Kamau [EMAIL PROTECTED]
wrote:
Thanks Roger, I find the second option more
interesting,
Anyone know where to find a Thai DID to ring in SIP to asterisk?
(probably Bangkok)
Chris Coulthurst
[EMAIL PROTECTED]
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To
Anyone know where to find a Thai DID to ring in SIP to asterisk?
(probably Bangkok)
Chris Coulthurst
[EMAIL PROTECTED]
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To
On Wednesday 20 July 2005 15:49, Angus Comber wrote:
Hello
I see it is possible to buy Flash Disks up to 4GB now. Has anyone any
experience of building an Asterisk system with a flash disk as the only
storage device? Any brands you recommend? Is 2 or 4GB enough for an
Asterisk
Anyone know where to find a Thai DID to ring in SIP to asterisk?
(probably Bangkok)
Chris Coulthurst
[EMAIL PROTECTED]
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To
On Wed, 2005-07-20 at 10:39 -0700, Victor Rini wrote:
David Stude wrote:
#2, I'm planning to interface Asterisk with a Norstar MICS via PRI. Can
anyone recommend a reference book or site more suited to this task?
Sorry that link is kind of dead.
I have the pdf if anyone is
Hello Adam,
In my opinion there should be only one transfer function, let suppose
it's called by #.
AG Wrong, which other phone system have you used where every time you try
AG and use some IVR that says Enter your xyz number followed by the # key
AG and you end up being interrupted by
On Thu, 2005-07-21 at 09:59 +0200, Alessio Focardi wrote:
PF Oh, you mean the completely natural feeling put them on hold, dial
PF new party, tell them you have a transfer, hit transfer? I want some of
PF whatever kool-aid the person who thought that one up had. I still feel
PF like I'm
hi all
i suggest to create a goup in hotmail in order to discuss any problem on line
in msn
i think it's more practical than e-mail group
Protek-on: CaraMail met en oeuvre un nouveau Concept de Sécurité Globale -
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On Thu, 2005-07-21 at 10:41 +, ali kia wrote:
hi all
i suggest to create a goup in hotmail in order to discuss any problem on line
in msn
i think it's more practical than e-mail group
If that serves you better than this list or the existing irc channel
(irc.freenode.net #asterisk) then
sometimes my tdm11p reads the caller id and sometimes
doesnt read it and give me this :
Jul 21 13:55:50 NOTICE[6284]: callerid.c:307
callerid_feed: Caller*ID failed checksum
Jul 21 13:55:54 WARNING[6284]: chan_zap.c:5739
ss_thread: CallerID returned with error on channel
'Zap/4-1'
what is the
We had an experience with asterisk in a 512 MB IDE Compact Flash card. It uses nearly 400 MB of storage with a minimum installation of Linux, and 2.6.10 Kernel. I has web access with AMP Portal with needed modules as apache, php, etc. But the size can be less.It works fine. You can search about
On Thursday 21 July 2005 12:41, ali kia wrote:
hi all
i suggest to create a goup in hotmail in order to discuss any problem on
line in msn i think it's more practical than e-mail group
also I would prefer not to switch to something M$ based...
___
Hello
I have downloaded asterisk-addons but when I make
install get:
cc -fPIC -I../asterisk -D_GNU_SOURCE
-DMYSQL_LOGUNIQUEID -I/usr/include/mysql -c -o
app_addon_sql_mysql.o app_addon_sql_mysql.capp_addon_sql_mysql.c:164:64:
macro "AST_LIST_REMOVE" requires 4 arguments, but only 3
without shattering what you are trying to do, asterisk wiki is the best
effort in existance im my opinion. IRC and mailing list.. I dont think
we need any more..
Keeping in mind im a nobody ;)
On 19/07/05, matt001 [EMAIL PROTECTED] wrote:
if it's of no use, we can always convert it for other type
hi all
i'm working in the asterisk pbx, my pbx manage good
outgoing calls but when i try to dial an incoming call i got this
message :
Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)...
but i could hear no thing on my diax,
i'm using TDM04B, and by my
Aldo Bergamini wrote:
The error is the same, afaik.
What I can't understand is why the make is entering in the directory '/
usr/src/linux-2.6.11.4-21.7-obj/i386/default'; I am by far not expert,
but I would expect it to go fiddle with a '586' directory.
Just a guess, your simlink is
Hi Angus,
I don't believe it can be the root password of mysql, I used to install
the addons without even haved installed mysql server yet, I guess we need
to know which platform are you working on and which version you are trying
to install.
Thx
MAG
Angus Comber wrote:
Hello
I have
I've built Asterisk from recent CVS sources on a Solaris 10 X86 box. I
tweaked the makefile to get the build to run using gcc. And most
recently ran into va_args problems with new code in asterisk/utils.c.
It seems to run OK and register with my VoIP provider, but I'm still
having trouble
On Thu, 2005-07-21 at 12:19 +0100, Angus Comber wrote:
Hello
I have downloaded asterisk-addons but when I make install get:
cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID
-I/usr/include/mysql -c -o app_addon_sql_mysql.o
app_addon_sql_mysql.c
app_addon_sql_mysql.c:164:64:
sometimes my tdm11p reads the caller id and sometimes
doesnt read it and give me this :
Jul 21 13:55:50 NOTICE[6284]: callerid.c:307
callerid_feed: Caller*ID failed checksum
Jul 21 13:55:54 WARNING[6284]: chan_zap.c:5739
ss_thread: CallerID returned with error on channel
'Zap/4-1'
hi
i would lke implement attended transfert (or consultative transfer) on
asterisk server,
but i don't find doc about this.
Could you help me with some doc about attended transfert?
thanks
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On Jul 21, 2005, at 12:00 AM, Vic wrote:Dear all, I had Tom Rymes and several others suggest how I can implement sending fax using Asterisk. The idea is to have On-Demand-Fax. Unfortunately, I wrote down the wrong workflow: the real one is: 1. Person will call our phone number 2. He will be
On Friday 15 July 2005 21:12, Mike M wrote:
I'm just trying to decide if the extra ?200 for the A104u is worth it :)
Isn't it the other way around? c u?
Yes you're quite right. I think I must have just taken the headstaggers last
Friday :)
Cheers,
Gavin.
1) send sound to the caller of an ongoing call
2) retain control so the call can be terminated based on a timer (or
whatever)
Any tips would be greatly appreciated! Thanks in advance.
Use the manager API to terminate the call if their credit reaches zero,
connect and process active
Hi,
I've configured a queue with dynamic agents and leastrecent strategy.
If the "least recent agent" doesn't pick up the current call from the
queue, the call will be presented to him again and again, even when
there's yet another agent available.
I would expect that after timeout occurs on
On Wednesday 20 July 2005 20:15, Eric Wieling aka ManxPower wrote:
As I understand it, adding VAD/Silence would require redesigning the
entire RTP stack of Asterisk.
My understanding is that with the new jitter buffer both of these things are
completely doable now since nothing's timed off the
Adam,
That's an interesting approach. I have a general question that arose
from your comment. You suggest using the h extension (or g option of
Dial) to reduce credit. What would happen if asterisk is restarted or
crashes with ongoing calls? Is there any trace of those calls in
order to
Dear Asterisk experts,
I've just downloaded Asterisk CVS version (since I'd like to manage
its configuration from RealTime).
Next, I have kphone on the same Linux host, and VmWare virtual
machine with Windows and X-Lite IP phone inside.
I successfully tested the demo's with X-Lite, but failed
Here is a snippet from my remote voicemail application where a user
needs to enter a code which is then matched against the db
;
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID}) ;just so I can see who
called, may wish to save sometime
;exten = s,4,noop()
exten
My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j
It is a version put together by Junghanns.net - for working with their ISDN
cards. Mmm I wonder if that is the problem? If so then what version of
asterisk-addons do I install. I didn't see anything about asterisk-addons
on the
Title: Dropping call
Hi, after upgrading from 1.0.7 to 1.0.9 I now seem to have a call drop problem. It mostly happens after about 1min 30 secs but also happens are random intervals. Everything was fine with 1.0.9 and I'm using the same config files. Could it be a zaptel problem? Does anyone
On Thursday 21 July 2005 15:28, Angus Comber wrote:
My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j
It is a version put together by Junghanns.net - for working with their ISDN
cards. Mmm I wonder if that is the problem? If so then what version of
asterisk-addons do I install. I
On Thu, 2005-07-21 at 09:22 -0400, Waldo Rubinstein wrote:
On Jul 21, 2005, at 9:04 AM, Eivind Trondsen wrote:
1) send sound to the caller of an ongoing call
2) retain control so the call can be terminated based on a timer (or
whatever)
Any tips would be greatly appreciated! Thanks in
On Thu, Jul 21, 2005 at 02:28:50PM +0100, Angus Comber wrote:
My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j
It is a version put together by Junghanns.net - for working with their ISDN
cards. Mmm I wonder if that is the problem? If so then what version of
asterisk-addons do
Dear friends,
Ihave a asterisk-1.0.9 verison with me in redhat linux 9.0
I am trying for ACD
I have two agents 1001,1002 and one queue called "queue1"
My requirement is like when ever any member try to enter into the queue
it should say messages like you are next and also hold time
the related
On Thu, 2005-07-21 at 14:28 +0100, Angus Comber wrote:
My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j
It is a version put together by Junghanns.net - for working with their ISDN
cards. Mmm I wonder if that is the problem? If so then what version of
asterisk-addons do I
On Thu, Jul 21, 2005 at 03:44:03PM +0200, Christoph Eicke wrote:
On Thursday 21 July 2005 15:28, Angus Comber wrote:
My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j
It is a version put together by Junghanns.net - for working with their ISDN
cards. Mmm I wonder if that is the
On 7/20/05, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Jul 18, 2005 at 05:44:17PM -0700, Geoff Karl wrote:
Being that my end goal is to stream an mp3 file any ideas on how this
should be configured.
Why stream an mp3 file in the first place? Is the network saurated? Do
you really
On Thu, 2005-07-21 at 13:00 +0900, Vic wrote:
Dear all,
I had Tom Rymes and several others suggest how I can implement sending
fax using Asterisk. The idea is to have On-Demand-Fax.
Unfortunately, I wrote down the wrong workflow: the real one is:
1. Person will call our phone
I dont know if is a common problem but what
Ive found:
First my config:
Zaptel.conf:
defaultzone=uk
fxoks=1-2
fxsks=3-4
loadzone = uk
Zapata.conf
[channels]
language=en
context=from-pstn
usedistinctiveringdetection=no
usecallerid=yes
cidsignalling=v23
cidstart=polarity
I've got several agents on a queue. However, they often forget to go
not ready or log off when they can't answer the phone.
I would like a person calling my queue to be on the queue for a max of 2
minutes, and I'm using the rrmemory strategy.
I put a timeout of 12 on the call to my agent in
Andrew Kohlsmith wrote:
On Wednesday 20 July 2005 20:15, Eric Wieling aka ManxPower wrote:
As I understand it, adding VAD/Silence would require redesigning the
entire RTP stack of Asterisk.
My understanding is that with the new jitter buffer both of these things are
completely doable now
Unfortunately, I do not have the correct pronounciations- but there
are some sounds missing in say.c, for at least Portuguese:
pt-ah.gsm
pt-ao.gsm
pt-de.gsm
pt-e.gsm
pt-ora.gsm
pt-meianoite.gsm
pt-meiodia.gsm
pt-sss.gsm
From what I can tell, they've been missing from the main repository
I actually had the same problem for a while. It would stop registering or it would say something about register timeout.
I have made two changes that have successfully resolve this issue for more then a week now.
I have added nat = yes in sip.conf under broadvoice peer section and I
have
Rob Engstrom wrote:
We've just setup our [EMAIL PROTECTED] server, with our quad port card.
Everything works
well so far.
One thing I notice is that when I leave the handset on the hook and dial a
#, all is well. If I pick up the phone and dial, it cuts off at 10 digits,
which is a
To answer my own question... the solution is to have both ends run the
same version.
Mark
Mark Willis wrote:
Two asterisk servers, one running a recent HEAD, the other 1.0.9. I
have both ends set up with trunk=yes, notransfer=yes, type=friend. I
notice that the trunking works from HEAD to
Has anyone attempted to change the MeetMe enter and exit sounds. I see
that the raw values in the enter.h and exit.h files. If I want to change
the sounds is it as easy as converting the auto files to .raw and place
the text in the file? I don't believe there is a header in the raw
format.
Thanks
Hello,
Does anybody have
the latest Boot ROMs for the IP500 and IP 600 Polycom
phones.
I have one of each
and can't find the Boot ROM v 3 anywhere to download.
I would also love
a good sample phone.cfg and sip.cfg files from an Aussie asterisk user to look
at.
Also the ip500 is
having
On Thursday 21 July 2005 10:32, Matthew Boehm wrote:
I figured timing could be done off a zap card or USB, just like with
meetme.
There's no need for a hardware timing source. The kernel timers are more than
adequate for 20ms.
-A.
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I didn't sign up with those yokels. Anyone else got spammed by them?
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Go ahead, create an MSN group.
You'll be very lonely over there.
-Original Message-
From: ali kia [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 21, 2005 5:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] a ne pas voir
hi all
i suggest
-Original Message-
From: Lee Howard [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 20, 2005 11:57 PM
Subject: Re: [Asterisk-Users] RE: Business Edition
Any consultant, business, or person that intends to reliably sustain
...
As for the dual-license issue... there are
Thanks for expanding on this. It is very clear now.WaldoOn Jul 21, 2005, at 9:47 AM, Adam Goryachev wrote:Well, I almost said it, but I figured by extrapolation people might work it out by themselves... Since you are checking the calls in progress on a regular basis, you might as well deduct the
Hi all,
Looking for some advice as to how best to configure a call queue situation.
Basically wondering whether it can be achieved with a standard asterisk
based queue or not?
I have calls coming in to several different DDIs, these would in theory all
route to same call queue as its important
Andrew Kohlsmith wrote:
On Wednesday 20 July 2005 20:15, Eric Wieling aka ManxPower wrote:
As I understand it, adding VAD/Silence would require redesigning the
entire RTP stack of Asterisk.
My understanding is that with the new jitter buffer both of these things are
completely doable now
Hi all,
I just upgraded from Asterisk 1.0RC1 to Asterisk 1.0.7 and our dtmf no longer
works with external phone systems. I have a Wildcard TDM400P with 4 FXO's?
(it connects to analog lines). No changes were made to the config files.
Here's my config:
/etc/zaptel.conf
fxsks=1-4
loadzone = us
attended transfer are implemented on some cases on the phone side, if
you need attended transfers on dial plan you need use asterisk CVS
HEAD, i are using asterisk CVS HEAD and attended transfer work very
well.
just install asterisk CVS HEAD and configure features.conf file,
on voip-info.org have
you can download amsn it work under linux i have it and it works succesfully
De: Christoph Eicke [EMAIL PROTECTED]
A: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Objet: Re: [Asterisk-Users] a ne pas voir
Date: Thu, 21 Jul 2005 13:13:59
Hi all!i'm in course to implement Faxing in my
asterisk box and for that I've installed all succesfully like libtiff and
spandsp and next rebuild and reinstall asterisk modules, but when i call to
Rxfax from dialplan nothing happens and i get some errors like "XCN with final
frame tag
Paul Hewlett wrote:
On Wednesday 20 July 2005 15:49, Angus Comber wrote:
Hello
I see it is possible to buy Flash Disks up to 4GB now. Has anyone any
experience of building an Asterisk system with a flash disk as the only
storage device? Any brands you recommend? Is 2 or 4GB enough for an
On Thu, 2005-07-21 at 15:56 +, ali kia wrote:
you can download amsn it work under linux i have it and it works succesfully
I don't think the software was the point, it was the hotmail part.
Never mind it would be a bit like Esperanto, if you can find the other
person who speaks it
Olle E. Johansson wrote:
Andrew Kohlsmith wrote:
On Wednesday 20 July 2005 20:15, Eric Wieling aka ManxPower wrote:
As I understand it, adding VAD/Silence would require redesigning the
entire RTP stack of Asterisk.
My understanding is that with the new jitter buffer both of
Olle E. Johansson wrote:
...when the new jitterbuffer is included and if it's enabled...
Please help us test the SIP/RTP jitterbuffer!
It's available in the bug tracker!
/Olle
post 'da bug number
-Matthew
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Hi,
My asterisk server crashed with that error message. I'm using 1.0.9. I
don't know how to replicate the problem (although this is the second
time the server crashes). I have two (~40M each) core files, but I don't
know how to debug.
any help is well appreciated.
Thank you.
How are these faxes arriving in your * server?
If you have a line handler card like the TE400 types then this should
not be a problem. If you are getting your faxes from a VoIP service like
Broadvoice then you should make sure that you are usling ULAW as the
codec. Anything else will
On Thu, 2005-07-21 at 15:56 +, ali kia wrote:
you can download amsn it work under linux i have it and it works succesfully
I think he was refering to the service provider, MSN as in MicroSoft Network,
as opposed to the operating system. There is already a large enough user base
and
Anyone know of a place to get a Thailand DID that will ring in to
asterisk in the US at a nice price?
Chris Coulthurst
[EMAIL PROTECTED]
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Hi,
On Wed, 2005-07-20 at 14:49 +0100, Angus Comber wrote:
I see it is possible to buy Flash Disks up to 4GB now. Has anyone any
experience of building an Asterisk system with a flash disk as the
only storage device? Any brands you recommend?
Beware that some flash producers sacrifice seek
For a long time now we've allowed people to publish a wide variety of
URI against their enum records such as SIP/IAX2/H323 for VoIP and other
types for non-VoIP such as HTTP/MAILTO etc.
For the most part these record types aren't listed or aren't utilised so
I've done up a quick hack for firefox
Anyone know of a place to get a Thailand DID that will ring in to
asterisk in the US at a nice price?
Chris Coulthurst
[EMAIL PROTECTED]
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Anybody know where to find Thailand DIDs that can ring in to my * in the
USA on SIP?
Oh, and a good price, too! ;)
Chris Coulthurst
[EMAIL PROTECTED]
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Matthew Boehm wrote:
Olle E. Johansson wrote:
...when the new jitterbuffer is included and if it's enabled...
Please help us test the SIP/RTP jitterbuffer!
It's available in the bug tracker!
/Olle
post 'da bug number
http://bugs.digium.com/view.php?id=3854
Download the patch,
Jay Milk wrote:
Who is getting the better end of the deal?
Well, Digium, of course. I certainly hope that they've made way more
money from Asterisk than I ever expect to save or make. And I certainly
expect that Digium has made way more money from Asterisk because they've
open-sourced
Please visit http://chan-sccp.berlios.de/
- New sccp.conf parser. You need to edit your old sccp.conf and update
it according to the new sccp.conf (conf/sccp.conf)
- Button template on phone 79[2467]0 has been improoved. Now you can
choose the line/speedial button position. 7914 can now use
All
I currently have asterisk setup at home
and everything seems to be working great running Asterisk v1.8 and several Iaxy
devices. Call waiting signals come through to alert the user of a call waiting
call but when using the flash button on the analog phone the current user is
placed
On Thu, Jul 21, 2005 at 10:41:57AM +, ali kia wrote:
hi all
i suggest to create a goup in hotmail in order to discuss any problem
on line in msn i think it's more practical than e-mail group
Create one, and announce it in the proper channels, if you like.
Frankly, I see no reason why you
Hello,
On Wed, 20 Jul 2005, Wiley Siler wrote:
For the fella who wanted MOH music
Royalty free stuff can be found here.. The Acoustic Guitar is a nice
collection...
http://www.freeplaymusic.com/
Cheers,
W
I spoke with Scott at freeplay today, who said that licensing is
Hi,
This is my setup;
1. PSTN == Cisco == Internet
== Asterisks == Grandstream Phone
2. Grandstream ATA =SIP
Proxy== Internet == Asterisks == Grandstream
Phone
In both cases above when I dialed the DID
(say) 1-213-444-1234 from either the PSTN or Grandstream ATA the response
I "see" on
On Thu, Jul 21, 2005 at 07:04:43AM -0700, Geoff Karl wrote:
On 7/20/05, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Jul 18, 2005 at 05:44:17PM -0700, Geoff Karl wrote:
Being that my end goal is to stream an mp3 file any ideas on how this
should be configured.
Why stream an mp3
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