Ok, i understand,
But i don't know how to get the IP Adress when a softphone is registred,
and how to send to this IP adress, and call number to the softphone, for
an incoming call.
best regards,
Olivier S
Anton Frolov a écrit :
it was not a real code, but just a schema.
I can't write a
On 11/29/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
You can have your own external script to do whatever you want when vm is
left
from voicemail.conf:
; If you need to have an external program, i.e. /usr/bin/myapp
; called when a voicemail is left, delivered, or your voicemailbox
; is
Hi,
This question is both technical and business related.
I've got a prospective customer in France which belongs to Hotel industry.
He has a lot of visitors coming from the Nederlands.
I'm studying the opportunity to offer phone services to those visitors.
The service I'm thinking about is
There was something screwy going on with kernel vs kernel-devel.
So I rolled back to kernel-*-2.6.9-42 rather than kernel-*-2.6.9-42.0.3.
Zaptel has now installed successfully. I don't believe this is a problem
with 2.6.9-42.0.3 per se. Rather my system had different versions of kernel
vs
Your script will have to read the extra info from the msg.txt files
or it's realtime equivalent.
M
Now, maybe I'm stupid but how exactly
do I get details to it regarding all those VM variables that are
inserted when the email is normally sent out from voicemail. You know
the VM_NAME,
You could of course edit app_voicemail.c to pass more info...
Round about line 2329:
if (!ast_strlen_zero(externnotify)) {
if (messagecount(ext_context, newvoicemails,
oldvoicemails)) {
ast_log(LOG_ERROR, "Problem in calculating
number of voicemail
I have a customer who wants their Asterisk system to play an announcement
to the caller while a DTMF string is being sent from the caller's phone.
As far as I am aware, this isn't possible: it's only possible to detect
DTMF if the first incoming digit interrupts the announcement.
Is that
I use BFD on several of my servers. Works great.
http://www.rfxnetworks.com/bfd.php
- Original Message -
From: Jeronimo Romero
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, November 28, 2006 11:54 PM
Subject: [asterisk-users] iptables example
Is there something similar or better than HUD pro out there for asterisk
PBX. HUD pro is wonderful thing, but they require complete Fonality product
to be purchased first, and don't sell it as a stand alone product. If
someone is not interested in Fonality product but is ready to purchase some
Its easy. In sip.conf, under each phone/DID setting, define a different
port. e.g. If 8789092323 is assigned to a person in country abc where SIP is
blocked, in your sip.conf, under [8789092323] add port=12000 or whatever you
like. This will override the default port=5060 setting in sip.conf.
I have seen the answer to this question previously, perhaps I am just not
asking the question correctly. For manager-based apps that do not explicitly
set a callerid is there anyway to overide the system default of asterisk
On 11/28/06, Tim Panton [EMAIL PROTECTED] wrote:
On 28 Nov 2006, at
For voicemail to email solution, just wanted to ask the experts, which one
is better and why: sendmail or postfix, or something other.
--
Zeeshan A Zakaria
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
On 11/29/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
You could of course edit app_voicemail.c to pass more info...
Round about line 2329:
if (!ast_strlen_zero(externnotify)) {
if (messagecount(ext_context, newvoicemails,
oldvoicemails)) {
JO == J Oquendo [EMAIL PROTECTED] writes:
JO One thing I noticed about Asterisk and group rings is, if a phone
JO is not registered but in the group, sometimes it won't ring.
What did you expect? If it isn't registered, Asterisk doesn't know how
to reach it, and therefore it doesn't ring.
On Wed, Nov 29, 2006 at 04:48:37AM -0500, Zeeshan Zakaria wrote:
For voicemail to email solution, just wanted to ask the experts, which one
is better and why: sendmail or postfix, or something other.
As far as carrying voicemail messages and delivering them on a system
that does not have a high
Hi,
I would advice you to buy some minutes at a (or several for redundancy) SIP
provider located in France, so you don't need anything in Netherlands... As
you have SDSL, connexion between you and the provider should be really good.
If you don't need incoming calls from Netherlands through SIP,
Hello,
I am trying to use chan_misdn on a junghanns QuadBRI card.
Using the latest install-misdn-mqueue from beronet, all installation
went well apparently. However when I try to load the card it is not
recognized:
# modprobe hfcmulti type=0x04 protocol=0x12,0x12,0x2,0x2
Hi,
During off hours, a server of mine simply forward incoming calls to an
outside number, so that no user is locally available to report or notify
downtimes.
As availability is here a major requirement, I'm looking for a cost
effective and reliable way to monitor this server.
Should I simply
On Wed, Nov 29, 2006 at 11:45:50AM +0100, Louis-David Mitterrand wrote:
Hello,
I am trying to use chan_misdn on a junghanns QuadBRI card.
Using the latest install-misdn-mqueue from beronet, all installation
went well apparently. However when I try to load the card it is not
recognized:
Does anyone have any ideas? I am pulling my hair out :-)
I changed email address's which is why the names different.
Thanks in advance
- Original Message -
From: Admin @ TheAdmiralNelson.Com
To: asterisk-users@lists.digium.com
Sent: Thursday, November 23, 2006 6:54 PM
Somewhere did I see a test script.
I will see if I can find it once more.
With that information should you be able to write a simple script that
monitor the server and then will notify you if the server stop responding.
PING wold maybe also be a help.
//Mattias
On 29/11/06, Olivier [EMAIL
Hi!
I have only used 7940 and 7905.
The 7940 are supporting TFTP and I did use that to upgrade them.
I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware
and then the on that I am using.
//Mattias
On 29/11/06, Paul [EMAIL PROTECTED] wrote:
Does anyone have any ideas?
I've just ordered a Siemens Gigaset C450 IP cordless
IP/DECT phone, given that it's supported by asterisk
http://www.voipuser.org/review_41.html
However, I see that a slightly better Gigaset S450 IP
is available for only a slight price premium.
Are there any user experiences with the S450 IP?
Hi Mattias,
That is what I did for my 7960 and what I need to do for this. However my
problem is when I un tar the cisco file it won't run. I think it needs call
manager :-(
Paul
- Original Message -
From: Mattias Andersson
To: Asterisk Users Mailing List - Non-Commercial
your client have to solicitate at the ISP the unblock of this port,i live in argentina, its a normal that happens, your client have to solicitate tue unblock of that port.
Saludos
Leonardo
FREE pop-up blocking with the new MSN Toolbar MSN Toolbar Get it now!
Hi,
I'm able to place outgoing calls using mISDN,
but I cannot get incoming calls to work.
Whenever someone calls one the incoming numbers I get this:
Nov 29 13:21:48 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log:
Extension can never match, so disconnecting
The caller is then informed by
I have one account which was created 3 weeks ago
and 1 that was created 2 days ago, neither work.
jason schreef:
last I had heard, pretty much all FWD accounts that were created in
the past year or so no longer work with IAX. Still don't know why.
Timothy Parez wrote:
I've got the same
Hi Paul!
I do thing you could use a TFTP bout I have not ben woring with that phone.
Could you post your TFTP loog?
//Mattias
On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote:
Hi Mattias,
That is what I did for my 7960 and what I need to do for this. However my
problem is when I un tar the
I just sent an e-mail to the FWD support address,
I'll let you know where that gets me.
jason schreef:
last I had heard, pretty much all FWD accounts that were created in
the past year or so no longer work with IAX. Still don't know why.
Timothy Parez wrote:
I've got the same problem here.
Hi Eugen,
Eugen Leitl wrote:
I've just ordered a Siemens Gigaset C450 IP cordless
IP/DECT phone, given that it's supported by asterisk
http://www.voipuser.org/review_41.html
However, I see that a slightly better Gigaset S450 IP
is available for only a slight price premium.
Are there any
Hello,
I'll try to explain better what i want to do:
1 - i've developped a softphone, with iaxclient.dll, in Windev Langage
(French langage - PC SOFT). This DLL doesn't work well with Windev, some
of pieces are ok. the ring, is not OK!!
2 - For detect the ring, i have make a listen server on
On Wed, Nov 29, 2006 at 01:49:28PM +0100, Florian Overkamp wrote:
Unfortunately I haven't yet had one in my hands, but from the
feature-list it seems a bit more value for money compared to the C450.
Especially being able to handle 4 SIP accounts/lines at one provider,
being able to add
Hi
Its not even at the tftp stage. When I run the image file from Chisco and
attempt to run setup I get a registry error. I am assuming its because its
expecting a call manager.
How do I upgrade the firmware? The image I have is only for callmanager
cmterm-7970_7971-sccp.7-0-2SR1
Anyone know
I am looking for a desktop control panel for zap (agent proxies). Does
any one know of an application that is similiar to a softphone but
controls zap/agents interfaces. I am looking for phone book, transfer,
and possibly presence control. And it must be standalone, unlike HUD pro
and hudLite.
Hi all,
is there a way I can put a line on hook ? I'd like to keep the line busy
on demand (es. dialing an extension will put on hook line n.1) so the
caller receives busy tone directly from PSTN and not from asterisk.
Thanks.
marco
___
--Bandwidth and
Hi believe that you nead a standalone image.
Would you consider use SIP image, that could be possible to find on the net.
//Mattias
On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote:
Hi
Its not even at the tftp stage. When I run the image file from Chisco and
attempt to run setup I get a
On Wed, 2006-11-29 at 13:26 +0100, Timothy Parez wrote:
Hi,
I'm able to place outgoing calls using mISDN,
but I cannot get incoming calls to work.
Whenever someone calls one the incoming numbers I get this:
Nov 29 13:21:48 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log:
Extension can
Just as an it works for me, I created a FWD account a couple of weeks
ago, which seems to be working fine. I am able to receive calls over
IAX2 via my IpKall number.
Jim
Timothy Parez wrote:
I have one account which was created 3 weeks ago and 1 that was
created 2 days ago, neither work.
Sorry to bother you all with what is probably a simple question.
I am attempting my first go at a simple AGI application using PHP (Getting
Asterisk to SAY PHONETIC ABC). I have dabbled with PHP but I am by no means
a professional standard developer.
My script seems to execute ok, and I
FWD works fine for me. I just set up a trunk in asterisk.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VOIP PBX) 1-866-638-1254
(Voip PBX) Free World DialUp: 780-217
WebSite: http://www.freeworlddialup.com/
We have Toll Free DID's instock
* * * NO
Hi Patrick,
it seems like Asteirsk cannot match any number, try uncomment the two
following lines:
;exten = _X.,1,Dial(SIP/timothy,30,r)
;exten = _X.,2,Hangup()
Giorgio Incantalupo
On Wed, 2006-11-29 at 13:26 +0100, Timothy Parez wrote:
Hi,
I'm able to place outgoing calls using mISDN,
I'm travelling today but I was just able to use Firefly to login to FWD via
IAX2. I called the echo test with no problems other than the lousy network in
this hotel.
My Astlinux server Also reports that it's registered with FWD via IAX2. My
account is a couple years old.
Michael
On Wed, 29
Thanks for all the help guys.
I cannot load the new SIP image straight on as the SCCP image is very old.
i read the FAQs posted on the lists and it tells me I need to upgrade the
SCCP image to at least 7 before I can load the SIP image.
This is the problem I am having. I cannot load SIP until
Derek Whitten wrote:
Norbert Zawodsky wrote:
RR wrote:
snip
Mate, I can't say it with authority but I'm almost certain that the
only DB that a specific driver was written for is MySQL. I think if
you use res_mysql.o you should be able to talk to mySql directly
without needing
Noah Miller wrote:
Hi Peder -
Is the storage of actual voicemail messages in a database still limited
to ODBC? If so, why?
Yes. Why? Nobody has developed a voicemail solution that directly
connects to a *SQL database for message storage.
A clear answer :-) Although a sad one :-(
Because
Hello list
We have a situation where calls need to be transfered to another extension.
We are using # to accomplish this but we found this is only working for
calls answered at the original called extension. If the call has been
forwarded to another extension or if the call has been picked up by
Still doesn't work for me.
Still get timeout
Michael Graves schreef:
I'm travelling today but I was just able to use Firefly to login to
FWD via IAX2. I called the echo test with no problems other than the
lousy network in this hotel.
My Astlinux server Also reports that it's registered with
Hi,
I did, that was my first try,
but it didn't work.
Giorgio Incantalupo schreef:
Hi Patrick,
it seems like Asteirsk cannot match any number, try uncomment the two
following lines:
;exten = _X.,1,Dial(SIP/timothy,30,r)
;exten = _X.,2,Hangup()
Giorgio Incantalupo
On Wed,
Hello list,
I am curious here if anybody here got an experience connecting Avaya to
Asterisk using H323 / T1. I am completely lack of experience with Avaya
and I wanna know if anybody here has connected Avaya to Asterisk using
H323 and managed to stabilize it. Google provides mixed comments
I get the following with debug on:
P[ 3] I IND :SETUP oad:497978546 dad:50556010 pid:10 state:none
P[ 3] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:
P[ 3] -- info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0
P[ 3] -- Bearer: Speech
P[ 3] -- Codec: Alaw
P[ 0] -- * NEW CHANNEL
jason wrote:
last I had heard, pretty much all FWD accounts that were created in the
past year or so no longer work with IAX. Still don't know why.
Timothy Parez wrote:
I've got the same problem here.
It can't register anymore -- timeout
Brian Capouch schreef:
I hadn't used FWD for
Hi,
Sorry, uncomenting that actually worked.
Now I need to filter on the last two numbers, that shoulnd't be to hard
I guess.
Tim.
Giorgio Incantalupo schreef:
Hi Patrick,
it seems like Asteirsk cannot match any number, try uncomment the two
following lines:
;exten =
Hi,
I've setup a trixbox 1.2.2 instalation with digium b410p. It uses misdn
driver.
I'm able to place calls and receive calls using this interface.
One thing that happens is when i dial an a number from a sip client to an
number that is routed through b410p and the called party rejects the
I've got to the point with FWD and IAX that I just connect directly via SIP
to IPKall, using my Asterisk box's address as the proxy. It simply works
better and eliminates another point of failure at FWD.
I also find it helps keep things a little more organized since I can assign
my own internal
Using this link
http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt
This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3
I get the following errors on make install
Any help would be GREAT!
Thanks
[CC] app_cepstral.c - app_cepstral.o
In file
Tomer Horn wrote:
Hello list,
I am curious here if anybody here got an experience connecting Avaya to
Asterisk using H323 / T1. I am completely lack of experience with Avaya
and I wanna know if anybody here has connected Avaya to Asterisk using
H323 and managed to stabilize it. Google
Setup:
Office A:
router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv
Asterisk: v.1.2.4
static IP
Office B:
router: Linksys WRT54GL running Linksys firmware v4.30.2
Asterisk: v.1.2.7.1
dynamic IP (using dyndns name)
Office A is set up with refresh dns and cron job for iax2 reload
Fran when you say specify the next hop do you mean the S0 line be an
extension in sip.conf or a context in extensions.conf?
Or should the line simply be tacked on to my [default] context?
Larry
Fran Oliveira wrote:
I think it is wrong. You should specify the next hop with some like this
On 11/20/06, Ralph Liebessohn [EMAIL PROTECTED] wrote:
On 11/20/06, Alex Robar [EMAIL PROTECTED] wrote:
Hi Ralph,
Have you setup your PAP2 to allow the 729 codec? I believe you actually
have to tell it that it's allowed to use that codec before it will work.
Cheers,
Alex
On 11/20/06,
I am attempting my first go at a simple AGI application using PHP (Getting
Asterisk to SAY PHONETIC ABC). I have dabbled with PHP but I am by no means
a professional standard developer.
Can't really say what is wrong with your code since I never did an AGI
in PHP without this class :
The Asterisk Manager Interface is driving me nuts.
Whoever wrote it should be drawn and quartered.
Sometimes the data comes back separated by \r\n, and sometimes it's separated
by \n.
The whole thing is completely inconsistent, and trying to write any kind of API
for it is -GHASTLY-
Doug.
That's not possible. These are residential people who hardly know enough
to hook up their PAP2 with detailed step-by-step instructions on hand and
support on the phone :)
Thanks,
Daniel
-Original Message-
From: Tom Lynn [EMAIL PROTECTED]
Sent: Wed, November 29, 2006 12:28 am
To: Asterisk
Wow. I didn't know you could do that. So, I could have something like this
in sip.conf:
bindport=5060,5080,5081,5082
and it will make Asterisk listen on all those 4 ports?
- Daniel
-Original Message-
From: Joseph [EMAIL PROTECTED]
Sent: Wed, November 29, 2006 1:31 am
To: [EMAIL
Ok. So, it's not that I can make Asterisk listen on multiple ports for any
SIP friend, but I could override the port on an individual SIP friend.
So, instead of having something like:
bindport=5060,5080,5081,5082
in the general section of sip.conf, I need to just have
bindport=5060
in the
Sometimes the data comes back separated by \r\n, and sometimes it's
separated by \n.
The whole thing is completely inconsistent, and trying to write any
kind
of API for it is -GHASTLY-
Doug,
What language(s) are you using? Just curious. I've been tinkering with
Perl, POE, and
Howdy,
Anybody have any ideas on how to record to a different file each time a call
is transferred by means of the transfer button on Polycom phones? I basically
need to be able to execute StopMonitor and an AGI script each time a call is
transferred without using features.conf for transfers.
Hall, Eric M. wrote:
Using
this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt
This
is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3
I
get the following errors on make install
Any
help would be GREAT!
Thanks
I am installling on a scratch asterisk running white box linux (fedora)
Does anyone know where to find them after the rpm runs. I am looking for
ircd and the perl dependancies. The instructions make a ton of
assumptions, so I am not sure what is happening here.
Jordan Novak
Senior
Douglas Garstang wrote:
The Asterisk Manager Interface is driving me nuts.
Whoever wrote it should be drawn and quartered.
Sometimes the data comes back separated by \r\n, and sometimes it's separated
by \n.
The whole thing is completely inconsistent, and trying to write any kind of API
for
Eric,
It looks like the definition for PTHREAD_MUTEX_RECURSIVE is within an
#ifdef __USE_UNIX98 (on Fedora Core 6, anyway). You could try defining
it within the Makefile. Similar to the _GNU_SOURCE definition in the
app_cepstral.so: app_cepstral.c stanza.
Bob...
On Wed, 2006-11-29 at 13:38
-Original Message-
From: Michael Collins [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 29, 2006 11:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
Sometimes the data comes back
We are thinking of setting up an Asterisk system to route calls between 2 of
our factories. Our idea is to connect an Asterisk box to each PBX and then use
SIP(or IAX) to truck between the 2 systems on our internal network.
I would be interested in any ideas regarding the connection points:
Dear List,
I'm looking for a coder/developer that can modify oh323 return codes on
asterisk
Example on based on SIP and h323.
Right now we are receiving :
Call Rejected (code 21)
Network Out of Order (code 38)
Need to able to replace dose codes with
- No Circuit/Channel Available (code 34)
On 11/29/06, Douglas Garstang [EMAIL PROTECTED] wrote:
The Asterisk Manager Interface is driving me nuts.
Whoever wrote it should be drawn and quartered.
You would need a ton of rope and a few hundred horses for that :)
The Manager API code is distributed across dozens of source files in
the
-Original Message-
From: Matt Florell [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 29, 2006 12:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What's up with the Manager Interface?!?!
On 11/29/06, Douglas Garstang [EMAIL
Hi Norbert -
I want to store all of my voicemail stuff in a database so that I can
give users web access to it, but I don't want to run web services on my
* server itself. If it is all in a DB, I can have a web box and a
separate SQL box and none of it should affect *.
Yes, you can do
Here's a good example. I'm trying to get SIP blf. I managed to split
my
result into a list of lines by splitting on ANY of \r\n, \n or \r. I
was
going use the column headings from the third line as my keys for my
dictionary/hash, rather than hard coding them. Notice anything? The
'Call
ID'
Hi Vincent -
Here's what I did on the X-Lite at home in the Topology section:
IP address : Discover global address
STUN server : Discover server
Port used on local computer : Manually specify range 8000-8019
Here are the ports that I forwarded from my NAT router at home:
UDP 5060
UDP 3478
On Wed, 2006-11-29 at 16:38 +0100, Timothy Parez wrote:
I get the following with debug on:
P[ 3] I IND :SETUP oad:497978546 dad:50556010 pid:10 state:none
P[ 3] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:
P[ 3] -- info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0
P[ 3] --
Hi Noah,
Noah Miller wrote:
Hi Norbert -
Just a thought: You could go the other way - share a volume on a
separate webserver, and have the asterisk box connect to the webserver
via NFS as a client, and store the voicemail on the NFS share. While
I don't have any exact numbers, it seems
I get an error when I do a make install
[EMAIL PROTECTED] app_swift-0.9.5]# make install
gcc -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC -DCHANNEL_HAS_CID
-DNEW_CONFIG -I/opt/swift/include -c -o app_swift.o app_swift.c
app_swift.c:49: warning: type defaults to `int' in declaration of
I thought I sent this out.. but don't see it so apologies if it
went already.
I am trying to get streaming MOH working but haven't been able to.. I
am running 1.2.x
Based on people's suggestions in other e-mails I've tried:
[scanner]
mode=custom
application=/usr/local/bin/mpg123 -s --mono -y
Hi List,
I have a Polycom 601 that when the user is on the phone they only hear one beep
and the CID of the second incoming call is not shown. Is there a way to have
the CID show up for the second call ? And a way to configure the phone to beep
more often if there is another call coming in. The
-Original Message-
From: Douglas Garstang
Sent: Wednesday, November 29, 2006 12:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
-Original Message-
From: Michael Collins
Thanks for the response!!!
I enabled debuging in the menuselect configuration for compiling
asterisk 1.4 beta3. In logging.conf enabled debug loggin to the
/var/log/asterisk/debug file and to the console. Restarted (not just
reload) asterisk and there is plenty of general debugging info in the
I'm wondering if anyone is having problems when
multiple users concurrently change their voicemail
passwords.
Consider the following scenario (based on
vm_change_password() in app_voicemail.c):
- user1 wishes to change his password so
voicemail.conf is opened and read into a buffer
- user1
On Wed, 29 Nov 2006, Douglas Garstang wrote:
G. Here's another example...
Action: Command
Command: sip show peer 2944093
Response: Follows
Privilege: Command
* Name : 2944093
Secret : Set
MD5Secret: Not set
Context : 180o_CallStart
Subscr.Cont. : 180o_WatchBLF
Doug,
Your issue isn't with the manager. It's with the CLI output you are
trying to hijack via manager :D If you run sip show peer 2944093 in the
CLI, you'll see a blank line, followed by a line that is * Name. It
appears what you really want is a manager Action to show a sip peer, in
which
If you have enough users where this comes up as a real issue, I'd recommend
migrating to Asterisk Realtime voicemail, then can have row-level locking etc.
if you use the right kind of storage engine... I've found problems using the
dial-by-name directory with realtime voicemail, but it seems
I have been using Enswitch. Has some bugs but over all works great. It's not
open source but worth the money.
- Original Message -
From: Guillermo Salas M. [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday,
I have asterisk 1.2.12.1 running with several client phone options. Our
echo cancellation is finally working great. The only problem I seem to
be having is there is background noise including beeping sounds at
regular intervals no matter which phone we use. Does anyone know why?
We are using a
-Original Message-
From: Steve Edwards [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 29, 2006 2:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
On Wed, 29 Nov 2006, Douglas Garstang
Anyone know if it posible to make voice promps native g726 or g711 format?
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In article [EMAIL PROTECTED],
Richard Lyman [EMAIL PROTECTED] wrote:
just wait till you get a 'hiccup' that causes a line to get cut off,
drop a char, and continue on next line. G
(examples below)
I've made heavy use of the Manager interface for over 2 years now, and
have never seen the kind
Hi all
For some dumb reason I decided to upgrade from Mandriva 2006 to 2007, thinking
I could install asterisk all over again. Anyway I did install asterisk, zaptel
and libpri. After install I ran modprobe zaptel which said zaptel not found.
Thanks to help on this mailing list I had a fix
-Original Message-
From: James Texter [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 29, 2006 3:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What's up with the Manager Interface?!?!
Doug,
Your issue isn't with the
phpldapadmin is pretty nice. I was using 2-3 different ldap clients to get
the job done until I got over my php bias and installed it. It lets me do
everything I want, without crashing.
On 11/27/06, Steven Baker [EMAIL PROTECTED] wrote:
Hello All,
we are using asterisk+openldap. Do is there
Greetings,
I am cutting my teeth with SIP phones and my first issue is getting a
Cisco 7940 to Authenticate with my VoIP provider (BBTelsys).
I did read some notes on the vo-ip website about 7.5 being the better
firmware version. Has anyone had trouble with 8.2 and SIP registering?
Should I
On 17:05, Wed 29 Nov 06, Scott Keagy wrote:
If you have enough users where this comes up as a real issue, I'd recommend
migrating to Asterisk Realtime voicemail, then can have row-level locking
etc. if you use the right kind of storage engine... I've found problems using
the dial-by-name
James Texter wrote:
Doug,
Your issue isn't with the manager. It's with the CLI output you are
trying to hijack via manager :D If you run sip show peer 2944093 in the
CLI, you'll see a blank line, followed by a line that is * Name. It
appears what you really want is a manager Action to
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