Hi,Here are the step by step instructions for setting up a brand new AudiocodesFXS gateway for use with an Asterisk server: Connect the gateway to a network switch and connect a computer to the same
switch. Then configure the IP address of the computer to 10.1.10.2. Then runyour web browser and
I am using Trixbox
1.0(Asterisk 1.2.7.1)at a customer site. They whishes to change the default
Voicemail behavior.
Standard
behavior
No answer/Busy -
send to Voicemail
Requested
behavior
No answer/Busy - message that if you press 9 you will
instead be cent to reception - send to
On Jun 27, 2006, at 12:19 PM, Yrving Rivas wrote:
Hi everybody:
Can somebody give me a hint?
I have tried with gsm files, with wav 8khz 16bits, wav 8khz
8bits...and no
way...what could be happening?
Try asking a question that makes some sense? What are you trying to do?
On Jun 27, 2006, at 8:24 PM, Richard Scobie wrote:
Tetsuya Yamamoto wrote:
I can't makel asterisk addon, asterisk-ooh323.
I use Asterisk and addons svn version.
The current svn version of asterisk has had the module loader code
redesigned and to date, the svn addons have not been updated to
Hi,
Has anyone successfully configured a HFC ISDN card with Singtel's ISDN-2
service? If so, can you share the settings required?
Thanks in advance.
KokMeng.
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On Jun 27, 2006, at 6:08 PM, shadowym wrote:
Hi there,
I am getting ready to set up a production Asterisk system. It needs
to be
stable. Upgrading, patching, rebooting, troubleshooting etc. are
pretty
much NOT an option once this thing is deployed. Like any phone
system, it
is
shadowym a écrit :
Hi there,
I am getting ready to set up a production Asterisk system. It needs to be
stable. Upgrading, patching, rebooting, troubleshooting etc. are pretty
much NOT an option once this thing is deployed. Like any phone system, it
is expected to just work.
Try
Since no know had an answer me I finally figure this out. It was a
corrupt sound file which was the source of this error. But I still don't
understand why asterisk crashes when you have a corrupt sound file!
Regards
Fredrik Jensen
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Same for me here, freebsd ports and same usage.
Running from months on a Dell 1850(biXeon 4Gb ram) with no problems.
Olivier
Jean-Michel Hiver a écrit :
shadowym a écrit :
Hi there,
I am getting ready to set up a production Asterisk system. It needs
to be
stable. Upgrading, patching,
Ok, on peut parler français alors ;)
Olivier
Jean-Michel Hiver a écrit :
Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig
Engels praten!
;-)
Pues my punto fue que un poquito de correo en otro idioma no hace
daño, y si ayuda mucho y molesta poco, ¿por qué quejarse?
Martin Joseph wrote:
Do you just mean the tar balls of 1.2.9 and latest addon?
Yes. I believe the svn addons package will be updated soon.
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Hallo.
I managed to configure asterisk to act as H.323 gateway using asterisk built in
support for H.323. I found it in ./channels/h323 directory of asterisk sources.
I wonder whether asterisk can play a role of H.323 gatekeeper. If Yes, could
You tell me some hints on how to do that.
How many channels have you guys been able to get with this?
The only problem I have with this is that it takes skype and a soundcard
(virtual or otherwise) and the API is really executing commands on a
running skype process. In my opinion its not worth it for 1 concurrent
call per account.
I
Well, look at it this way: if you get the working, you can buy one of those
tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard
and a ethernet port. Run Linux off a CF card and have it setup to *only*
interface with Skype and Asterisk. Basically, make a Skype ATA, but
Pawel wrote:
I wonder whether asterisk can play a role of H.323 gatekeeper
Not today. Although, disclaimed patches are gladly accepted at
http://bugs.digium.com.
Jeremy McNamara
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Hi Tom,
Thank you for your interest in my problem, I really am
desperate about this thing...
I have tried several versions one after another, and now
I'm using the one released on 04.07.2006 (SIP release
2.2.2).
Thanks,
Silviu
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi Tom,
Thank you for your interest in my problem, I really am
desperate about this thing...
I have tried several versions one after another, and now
I'm using the one released on 04.07.2006 (SIP release
2.2.2).
Thanks,
Silviu
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
On Wed, June 28, 2006 10:14, [EMAIL PROTECTED] said:
Well, look at it this way: if you get the working, you can buy one of
those
tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia
soundcard
and a ethernet port. Run Linux off a CF card and have it setup to *only*
interface
On 6/27/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
On Tue, 2006-06-27 at 15:00 +0200, Morten Isaksen wrote: Hi! I have this setup:
PABX --ISDN30-- Asterisk 1 --SIP-- Asterisk 2 --ISDN30-- TELCO Digium TE410P is used in both Asterisk 1 and 2. When I set the CLIR bit on the PABX the
On Wed, 2006-06-28 at 08:14 +, [EMAIL PROTECTED] wrote:
Well, look at it this way: if you get the working, you can buy one of those
tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard
and a ethernet port. Run Linux off a CF card and have it setup to *only*
KokMeng Loh wrote:
Hi,
Has anyone successfully configured a HFC ISDN card with Singtel's
ISDN-2 service? If so, can you share the settings required?
The AVM Fritz! PCI works fine with chan_capi for Singtel. At that time
(about 2 years back), I didn't test HFC because the driver was very
Hi guys,
I'm getting the following error when trying to compile zaptel on a
debian machine running 2.4.27-3-386.
gcc -g -c -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE
-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -I. -Iinclude
-O4 -g -Wall -DBUILDING_TONEZONE
Hi Guys
I have just installed a call center onto Suse 10. I have managed to do a
DBget (astdb) and extract the DNID numbers to play a DNID specific
greeting. We have installed Snom 320 and the customer would like us to
Send the DNID(nam) to the phone screens so that the agent will be able
to
Hi,
On 11:31, Wed 28 Jun 06, Terry Wade wrote:
Hi Guys
[snip]
I have tried to do a:
exten = _X.,1,Set(CALLERID(name)=${DB(CIDNAMES/${DNID})})
but this has not seemed to help.
Use this: exten = _X.,1,LookupCIDName()
That works great in my setup
--
Michiel van Baak
[EMAIL PROTECTED]
Hi,
It's kind of off-topic , but still within Asterisk. I developed an asterisk
module that send an
authentication to a radius server for call authorization and process its reply
(limited to
User-Name and Cisco or Quintum VSA h323 attribute). My question, is when it
make sense to use or
On Wed, Jun 28, 2006 at 08:14:56AM -, [EMAIL PROTECTED] wrote:
Well, look at it this way: if you get the working, you can buy one of those
tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard
and a ethernet port. Run Linux off a CF card and have it setup to *only*
On Wed, Jun 28, 2006 at 05:22:17PM +0800, Mark Davies wrote:
Hi guys,
I'm getting the following error when trying to compile zaptel on a
debian machine running 2.4.27-3-386.
What kernel-headers/kernel-source?
I hope you didn't extract a source tarball of 2.4.27/debian and linked
it to
On Wed, 2006-06-28 at 13:15 +0300, Tzafrir Cohen wrote:
Since you can make a Skype account for free and
can (for right now) make US and Canada LD calls for free, I think the cost
and time to make them would be worth it. :) And if you figure out a good
price for them, people might even
Hi All.
Somebody of you already passed below for this error?
Jun 28 02:25:03 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jun 28 02:52:08 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
Hi all,
I am looking for a developer or developers that can implement the following:
- Modify an Asterisk server in order to support one inbound RTP and
several outbound RTPs, I was thinking SEMS may provide a very good
starting point. The idea is to make a PA system over IP. We do *not*
I am using a2billing as billing system on tixbox but I have
a problem since the user call the destination number ,the ivr tell him about
him amount and ask him to enter the destination number ,my question is how can
I let the user call the destination directly .
Regards
Hi,
Is it illegal to use Uplink Skype2Sip software to connect a skype
account to a homepbx asterisk? ( Just to know... i don't want to be
bored because of asteriskpt.blogspot)
On 6/28/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
On Wed, 2006-06-28 at 13:15 +0300, Tzafrir Cohen wrote:
I am trying to upload a wav file to my asterisk through the AMP. That file
is going to be used as the receptionist voice.
The amp requires the file to be sampled to 8khz 16 bit which is done.
I am using the Freepbx portal to upload the file.
That file comes from a record I have paid for to a
Hi,When you originate a call asterisk essentially callouts to the
Specified channel and the when answers connects the the
context,extension,priority. What if I want my dial plan to make the
origination call and the destination call. What would I specify for my
dialplan/callout file?thanks in
Hi,you check your a2billing.conf file:; Please enter here the file you want to play when we prompt the calling party to enter his destination number; file_conf_enter_destination = prepaid-enter-number-u-calling-1-or-011
file_conf_enter_destination = prepaid-enter-destI think this file should help
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Or a TDM2400 with 4 FXO modules... (4x4=16) :)
Lito Lampitoc wrote:
oh sorry, 2 TDM400P with 4 FXO modules each :=)
On 6/28/06, *Lito Lampitoc* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
or TDM400P with four FXO modules perhaps?
On
Hi,
Take a look here: http://www.asteriskguru.com/tutorials/hdlc_bad_fcs.htmlit
might help.
Otherwise you can also try different settings for the
"span" line in zaptel.conf
Silviu
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josué
ContiSent: 28 June 2006 12:33To:
I love the added apps installed with trixbox, ARI, Web-Meetme,
FOP, and Reports are great. FreePBX on the other hand, is nearly impossible to
do everything with. Trying to edit the configs manually proves impossible due
to the excessive use of includes and macros. It is kind of like
This works great, however, when I look at the full log, it says that
the sendmail is executing prior to vm-audio. Any way to change this?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cullin J.
Wible
Sent: Tuesday, June 27, 2006 8:41 PM
To: [EMAIL
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA256
I have an installation where I'll have a site to site data DS1 for use between
two corporate offices. We'll have one asterisk server at each office. I'd
like to be able to route calls over the 24 channels on that DS1 between the
offices, instead
On 07:23, Wed 28 Jun 06, Jordan Novak wrote:
I love the added apps installed with trixbox, ARI, Web-Meetme, FOP, and
Reports are great. FreePBX on the other hand, is nearly impossible to do
everything with. Trying to edit the configs manually proves impossible
due to the excessive use of
Hi Vincent -
Sorry for the long delay in responding. I didn't see you message
until now due to the postfix problems on the mailing list. Anyway, I
see some clues here:
exten = s,1,Answer
exten = s,2,Waitexten(10)
exten = 100,Dial(Zap/2/014XX)
Then call in and after you're connected,
Hi,
I have a 100P card but even though there's no incoming route, it answers
the line after 2 to 3 rings. If I do create an incoming route, the same
happens, but it never rings the ring group or extension I enter. It's
almost as if the card acts as a modem. The caller hears nothing, just
I can confirm this.
AMP/TrixBox is a wonderful project but if you like to tweak
something or you became a moreexperienced user, it will became
soonas a straitjacket.
I'm still struggling to clean AMP config files to work with
a plain Asterisk install.
From: [EMAIL PROTECTED]
On Wednesday 28 June 2006 08:48, Jonathan Miller wrote:
An alternative is to put a router and switch at each end and extend a data
network to the other site for SIP traffic. Would that result in better
quality calls?
If you can ensure that voice traffic has top priority in all the routers
Neil Cherry wrote:
[snip]
How did you get access to the web config? What user and is it
the default password/access code?
type it's IP address into a web browser. Username: admin, password: psw
is the default.
cheers,
Paul.
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I installed an asterisk server with oh323 channel driver support.
Then I uploaded the H323 firmware on a AT320 phone (Usually I use it as a
sip phone, but I am using it just for test)
Let's say that I assigned 945 as phone number, account and password to this
phone, and its ip address were
Hi Jonathan -
I have an installation where I'll have a site to site data DS1 for use between
two corporate offices. We'll have one asterisk server at each office. I'd
like to be able to route calls over the 24 channels on that DS1 between the
offices, instead of over the voiceT at each location
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA256
I have a true leased line (a T1) between the two sites.
What parts do I configure for Asterisk to utilized the link bi-directional?
On Wednesday 28 June 2006 09:09, Andrew Kohlsmith wrote:
On Wednesday 28 June 2006 08:48, Jonathan Miller
is the following zaptel.conf configuration correct for TDMoE used for
pri-cpe signalling - is this possible at all ?
I couldn't find an example...
loadzone=nl
defaultzone=nl
# pri E1 card
span=1,1,3,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31
# hfc-pci 1
span=2,1,3,ccs,ami
bchan=32-33
dchan=34
#
Do you have more than one call per line enabled on the Poly? Is it
the phone or asterisk returning the busy? What does the console say?
On Jun 27, 2006, at 5:29 PM, Mike Staver wrote:
I have one extension setup for each Polycom 501 I have, and when I
try to call out on a conference call, I
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
What kind of T1? TDM? Data? What type of signaling are you planning
to use em? There is a lot of information that that question is
lacking for anyone to advise you ...
Jonathan Miller wrote:
I have a true leased line (a T1) between the two
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA256
Your response leads me to further question this setup...
It's a full data T that is not provisioned.
Being that I control the termination at each end, do I get to specify the
encoding?
On Wednesday 28 June 2006 10:17, Sean Cook wrote:
What
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Typically with a data t1 you are running either HDLC or PPP on either
end. I assume you have a cisco router on either end? Or are you
planning to plug asterisk with a Digium/Sangoma/Other T1 card?
Personally if it is a data t1 I would use a cisco
Oliver Vermeulen wrote:
JAMAICA DID'S - 1-876
NOW ACTIVE ON www.didx.org
Oliver Vermeulen
World Venture Group Telecom
Corporate Address:
147 New Haven Point Lane
West Palm Beach , FL , Miami
USA DID: +1 (305)722-1457
BE DID: +(32)9-395-5620
UK DID: +(44)870-478-8896
SIP : [EMAIL
Assuming it is a dedicated private line p2p T1
Assuming that 23 calls at one time is sufficient
Install a T1 card in each server, plug the T1 in and set one end ofr
pri net, the other for pri cpe.
zaptel.conf and zapata.conf are the files you are looking for. Just
define the 23
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
It always helps to read the original post... so I apologize. I think
what you are looking to do is route the calls over the existing data
t1 in which case all you need to do is create an IAX trunk between the
two asterisk servers addressing their
JAMAICA DID'S - 1-876
NOW ACTIVE ON www.didx.org
Oliver Vermeulen
World Venture Group Telecom
Corporate Address:
147 New Haven Point Lane
West Palm Beach , FL , Miami
USA DID: +1 (305)722-1457
BE DID: +(32)9-395-5620
UK DID: +(44)870-478-8896
SIP : [EMAIL PROTECTED] mailto:[EMAIL
Hi,
when attempting to dial an invalid number with Nikotel this is returned:
SIP/2.0 400 Bad Request request uri is neighter my realm or a valid dns
and Asterisk prints smth similar on the CLI. However it appears that I
cannot get access to 400 Bad Request from the dialplan because this
Hi Leo,
How stupid of me! I just realized that I needed a NT-1 box between my
HFC card and the ISDN line! What I observed was that the line was always
not active. Thanks for your reply anyway.
-kokmeng.
Leo Ann Boon wrote:
KokMeng Loh wrote:
Hi,
Has anyone successfully configured a HFC
Maybe one of you can help me with this:
We have T1's that come from both MCI and Global Crossing as uses
channelized (24
Ports per T) with inband (DTMF) ANI and DNIS delivery (format =
*DNIS*ANI*).
My old equipment was set for D4, AMI, SF and Wink Start and so is
Asterisk Server.
I've
Hi,
We have this set up in a few places.
Basically, what you need to do is play a sound file using the
Background() command. This allows for user entry. The sound file will
say press 9 for reception, or stay on the line to leave a voicemail.
This will go in the no answer/busy priority of your
Hello,
I have installed FreeRadius server on Trixbox Server. My problem is mysql is
not letting FreeRadius to login either locally or remotely. I also insert
proper entries in HOST and USERS tables. But it does not work I always get
ERROR 1045 (28000); Access Denied for user 'root'@'localhost'
Guys.
Ive seen on my asterisk messages log that asterisk has shutdown itself about
12 times in 5 days... The logs show nothing but:
[Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Drop call
[Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Release call
[Jun 28 09:40:02] VERBOSE[3172]: [Jun 28
Hey everybody,
Is it alright to run two TDM400s on the same machine?
If it is, how would one differentiate between the channels on each card?
So, if Im running strait FXS and my first card is fxsks 1-4, would the
second be fxsks 5-8? Would there be any interrupt problems?
Any help
Hi all
I have downloaded asterisk 1.2.8
try to make on RHEL AS 4
i get the following error
any clue
make[1]: Entering directory `/root/all/asterisk-1.2.8/res'make[1]: Nothing to be done for `all'.make[1]: Leaving directory `/root/all/asterisk-1.2.8/res'make[1]: Entering directory
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The only issues you could potentially run into is if all the modules
are FXS and they all needed to ring simultaneously... your power
supply may not be suited to handle to voltage requirements.
Sean
Ninneman, Tj wrote:
!-- /* Style Definitions */
Hello,
Here is a breakdown of the issue I am experiencing. I have three remote
employees, in various states, who have Polycom 501 phones. They are
unable to receive incoming calls after a few minutes of the phones being
plugged in. They work immediately after being plugged in, but they lose
the
Good catch - I hadn't realized that.
You are correct that in app_voicemail.c sendmail is run prior to the
externnotify script.
I see a few options: 1) change the code in app_voicemail.c 2) Use the
externotify script to assemble and send the email messages 3) Run a web
server and include a link
This isn't anything asterisk is causing.
Sounds to me like FreeRadius is not properly authenticating with mysql.
Example:
If I wanted to log into a mysql box remotely, it could be down like this
$ mysql -p -h XXX.XXX.XXX.XXX -u username databaseName
Which means that 'username' better be in the
We run them with 1 call per line, but when we first set them up they would
do 8. The problem was switching between calls on a single line. At that
time, however, the phone did not return busy and allowed the calls to stack
up.
This is set in the XML configuration files.
Cullin
-Original
As pointed out, just build your own system. If you understand the Freepbx dialplan, you can usually do almost anything you want in _custom files including redefining contexts in such a way that upgrades do not wipe them out. It's simply a matter of spending some time to see what is being done and
how can i make
exten = 780836,1,Dial(CONSOLE/dspSIP/,40,m)
the CONSOLE (ALSA) not to accepts the call always?
Juergen
alsa.conf is autoanswer=no
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To
You have to lower the registration interval in the phones to under a
minute otherwise the NAT hole closes and no calls come in.
Polycom has said that they are going to be putting in a keep alive in
the firmware at some point.
On 6/28/06, Von L. [EMAIL PROTECTED] wrote:
Hello,
Here is a
On 12:04, Wed 28 Jun 06, Von L. wrote:
Hello,
;_
;sip.conf
;_
[general]
port=5060
bindaddr=0.0.0.0
externip=XXX.XXX.XXX.XXX
localnet=XXX.XXX.XXX.XXX/255.255.255.248
BJ,
One other thing, did I need to have a version of asterisk already installed
before your version?
I had a blank system with Debian
installed lipri-1.2.3 (make clean, make, make install)
installed zaptel-1.2.6 (as above)
done svn checkout http:..functions asterisk-polycom
cd into
On Jun 28, 2006, at 4:45 AM, Yrving Rivas wrote:
I am trying to upload a wav file to my asterisk through the AMP. That
file
is going to be used as the receptionist voice.
The amp requires the file to be sampled to 8khz 16 bit which is done.
I am using the Freepbx portal to upload the file.
The Polycom's need to have their registration time lowered. Set it to 60 seconds which will re-register every 30 seconds. The polycom doesn't have any sort of 'keep alive' feature to keep the NAT holes open. There is information on the wiki fruther describing this and how to set it up if you don't
Von L. wrote:
Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running on
I would suggest you
Hi, What asterisk release (stable or dev) has support for a softphone
like Xlite (free) that uses h263 for video codec? (audio works fine)
in sip.conf I added
[xlite1]
videosupport=yes
allow=h263
allow=gsm
nat=yes
canreinvite=no
Also, what (proven/tested) hardphones with video support can be
Anton Krall wrote:
Guys.
Ive seen on my asterisk messages log that asterisk has shutdown itself about
12 times in 5 days... The logs show nothing but:
What version?
I'm running 1.2.9.1 and saw one of my Asterisk process, this morning,
just shut down for no apparent reason. I didn't have
Hi,Is there a way to by-pass the absence of ztdummy on a Debian powered Intel Macmini platform ?Regards
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FYI, when we had NAT routers at both locations setting qualify=yes
did not work.
On 6/28/06, Michiel van Baak [EMAIL PROTECTED] wrote:
On 12:04, Wed 28 Jun 06, Von L. wrote:
Hello,
;_
;sip.conf
Hi folks!Based upon your experience on the field what wifi sip phone would youreccomend ?A customer asked for a wireless * install and I'm looking for advice, tnxAlessio Focardi[[*] - Interconnessioni Italy
___
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Why use an application like sox - when you can make the voicemail
application do it natively:
exten = s,1,Dial(SIP/100,10)
exten = s,2,Voicemail([EMAIL PROTECTED]|ug(10))
The key is the g(10) parameter:
From the 'show application voicemail':
g(#) - Use the specified amount of gain when
-Original Message-
From: Michael Collins [mailto:[EMAIL PROTECTED]
Sent: Monday, June 26, 2006 4:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Not Able to
HandleComplexFailoverSituations
I get annoyed Stephen when Digium
At 09:57 PM 6/27/2006, you wrote:
2 TDM400P's with 4 FXO modules each = 8 FXO's = 8 PSTN lines.
It's like John said.
Very simple maths one would of thought, unless I'm completely off the mark.
In which case I do apologise.
Probably better off with the TDM2400, 2 fxo boards and the echo can.
Mike Puchol wrote:
Hi all,
I am looking for a developer or developers that can implement the
following:
- Modify an Asterisk server in order to support one inbound RTP and
several outbound RTPs, I was thinking SEMS may provide a very good
starting point. The idea is to make a PA system
Von L. wrote:
plugged in. They work immediately after being plugged in, but they lose
the ability shortly thereafter. They can always make outbound calls, but
only to real phone numbers, not extensions.
They each have NAT routers, and I have triple checked that they have
opened/forwarded the
Von L. wrote:
plugged in. They work immediately after being plugged in, but they lose
the ability shortly thereafter. They can always make outbound calls, but
only to real phone numbers, not extensions.
They each have NAT routers, and I have triple checked that they have
opened/forwarded the
asterisk:~# apt-get install kernel-headers-`uname -r`
Reading Package Lists... Done
Building Dependency Tree... Done
kernel-headers-2.4.27-2-386 is already the newest version.
Tzafrir Cohen wrote:
On Wed, Jun 28, 2006 at 05:22:17PM +0800, Mark Davies wrote:
Hi guys,
I'm getting the
When I fail to pick up a call from Asterisk to the PSTN to my cell
phone and let it go to voicemail, the sound quality is always really
bad. When I call my cell phone's voicemail a few minutes later, it's
really garbledy and sounds clipped or something.
I've tried using Monitor to record the
Try shutting off asterisk/zaptel and unloading any zaptel modules
(rmmod zaptel, wcfxo, etc) before doing the make install, so udev removes
any /dev/ entries associated with them (ie /dev/zap/transcode). If not
using udev/devfs, then perhaps unload all zaptel modules, rm -fr /dev/zap,
then
We've got a white route w/ VSNL. $0.09 / min, billing is 1/1 prepaid only.
SIP or H323 w/ G729 Codec. E-mail me off-list for testing.
Thanks,
Jon
- Original Message -
From: Jerry Romney [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; asterisk-biz@lists.digium.com;
I've been noticing lately what seems to be some distortian in the standard
asterisk sound files, used for voicemail. These files are stored on the local
Asterisk system. When Asterisk plays them, I can hear some cracles and pops.
I'd never noticed these until recently.
I did a little test.
Hi,
I used a FXO-Gateway to connect my VoIP to PSTN, using inband mode for
DTMF. It can work properly if I use this dialplan:
exten= 100.,1,Dial(SIP/xxx.xxx.xxx.xxx/111)
111 is a line number in the gateway.
but, I can't get the PSTN number that the caller dialed from the gateway.
So I tried
On Thu, Jun 29, 2006 at 01:32:36AM +0800, Mark Davies wrote:
asterisk:~# apt-get install kernel-headers-`uname -r`
Reading Package Lists... Done
Building Dependency Tree... Done
kernel-headers-2.4.27-2-386 is already the newest version.
2.4.27-*2*-386?
Tzafrir Cohen wrote:
On
On Wed, Jun 28, 2006 at 06:52:11PM +0200, Olivier wrote:
Hi,
Is there a way to by-pass the absence of ztdummy on a Debian powered Intel
Macmini platform ?
The absense of USB?
Use kernel 2.6?
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
Douglas Garstang wrote:
I've been noticing lately what seems to be some distortian in the standard
asterisk sound files, used for voicemail. These files are stored on the local
Asterisk system. When Asterisk plays them, I can hear some cracles and pops.
I'd never noticed these until recently.
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 28, 2006 12:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Standard Sound Files Distortion
Douglas Garstang wrote:
I've been noticing lately what
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