and is well known due to the fact u don't have a
precise clock source for meetme..
You need to have chan_dahdi dummie...
Hope it helps.
Marco Mouta
Enviada do dispositivo sem fios BlackBerry®
-Original Message-
From: Jeff Brower jbro...@signalogic.com
Date: Wed, 24 Feb 2010 18:25:07
of its span in /proc/dahdi file for a source: in the
description. Or even run:
strings dahdi.ko | grep source:
--
Marco Mouta
On Thu, Feb 25, 2010 at 8:15 AM, marco.mo...@gmail.com wrote:
It looks to me that u are having clock synchronism problems due to the fact
you are using Virtual Machine so
, but I believe this is only the small beginning….
Looking forward to hearing from you guys ;)
Cheers,
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, but I believe this is only the small beginning….
Looking forward to hearing from you guys ;)
Cheers,
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,
Steve Totaro
On Sun, Mar 15, 2009 at 9:20 PM, Marco Mouta marco.mo...@gmail.com wrote:
Hi,
problem is that you are saying that phone in sip.conf is at the same
ip address of your asterisk box so you are dialing into a loop to your
self asterisk box
[phone]
type=friend
context=phone1
,Dial(SIP/phone,10)
exten = s,2,Voicemail(line)
exten = s,3,Hangup
hope it helps.
don't forget to asterisk reload on cli.
Looking forward to hearing from you.
cheers
--
Marco Mouta
On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal vad...@gmail.com wrote:
Hi I looked at few emails related
try to set in your zapata.conf
overlapdial=yes
then in your asterisk cli
reload chan_zap.so
--
Marco Mouta
On Fri, Feb 13, 2009 at 9:21 AM, joek...@gmail.com wrote:
Default FreePBX context,
[from-pstn]
include = from-pstn-custom ; create this context
Hello,
please read bellow:
On Tue, Sep 9, 2008 at 11:04 PM, Christian Victor
[EMAIL PROTECTED] wrote:
Hi Asterisk users!
I have a little problem with an Asterisk 1.4.22 installation for a
customer. The PBX is connected to an E1 line and we have a few snom 300
attached to it.
The goal is
May be I'm wrong but:*
timeout - the maximum time, in seconds, the call will wait in the queue.
When this time expires, the next extension, by priority, will be executed.
By default the timeout is set to 300 seconds.
So you clearly have two ways to feed your database with your statistics:
If
Your solution is Asterisk Manager Interface
http://www.voip-info.org/wiki-Asterisk+manager+API
On Tue, Mar 18, 2008 at 6:24 AM, Janu Mukherjee [EMAIL PROTECTED]
wrote:
Hi,
I have our software with SIP running on it.I configured asterisk server as
proxy. How do I implement the call screening
I would recommend you Asterisk for Voice and Video and XMPP for Chat.
Asterisk in parallel with Jabberd2 (XMPP server) may feet your requirements,
and if you use a XMPP MSN Transport Gateway you can do even more.
On Mon, Mar 17, 2008 at 5:50 PM, Carlos Carvalhar
[EMAIL PROTECTED] wrote:
Dear all,
I've created a digium certified asterisk professional - dCAP linkedin
group for anyone, dCAP, interested:
http://www.linkedin.com/e/gis/60298/39AE1350DBF3
Best regards,
Marco Mouta
dCAP
November 2006
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If any of you around the world is aware of this values for VoIP SLAs I
would be thankful to exchange and discuss this info.
Thanks in advance.
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Marco Mouta
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if user
does hangup his/her call then message should be recorded
otherwise(after timeout) message is discarded. Is there any thing that
will help me...???
currently I am doing the same thing on pressing 1 with php agi script
and its working fine.
On 12/19/07, Marco Mouta [EMAIL PROTECTED] wrote
Post:
Asterisk CLI : sip show peers
Asterisk CLI : zap show channels
Asterisk CLI: zap show status
As well as your extensions.conf
Are you able to ping you GSM gateway? is connected via SIP or Telephony
interface card?
Best regards,
Mouta
On Dec 18, 2007 10:47 AM, Lolu Gbenga [EMAIL
In
http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home
is said Kerry Garrison that:
Both trixbox and FreePBX have phone-home mechanisms in them.
So does FreePBX phones home too?
On Dec 17, 2007 4:27 AM, Than Taro [EMAIL PROTECTED] wrote:
As I pointed out here
Thanks Tzafrir!
I really appreciate Free PBX.
Keep on going your good job.
Best regards,
Mouta
On Dec 18, 2007 11:59 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Dec 18, 2007 at 11:38:03AM +, Marco Mouta wrote:
In
http://www.trixbox.org/forums/trixbox-forums/open-discussion
What do you mean with record a call on hangup? If the calling party ends the
call you want to keep recorded file?
On Dec 18, 2007 6:27 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote:
Hello everyone out there, I am having a problem in call recording with php
agi library. I have already recorded
:= INTEGER in the range 1 to 100
best regards,
Marco Mouta
On Dec 12, 2007 8:08 AM, Eric Delaporte [EMAIL PROTECTED] wrote:
Hi @ all,
i set a server to a costumer of mine with a TE207P for use with 2 E1
Lines.
I set them together into one group in zaptel/zapata.conf
The point
modules.confthat I needed to copy from the backup
/usr/lib/asterisk/modules and give
the right permissions.
Am I missing something?
best regards,
Marco Mouta
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, making this multiple
instances try to access same asterisk channel (leading us to Avoiding
deadlock messages) ?
I mean applying the patch might solve the problems instead off all system
upgrade?
Best regards,
Marco Mouta
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regards,
Marco Mouta
On Dec 10, 2007 12:24 PM, Kovář Jan [EMAIL PROTECTED] wrote:
Hello.
I am going through the documentation and trying to find if asterisk can
help me in my case. It is quite difficult to find answer because I do not
know the exact question.
I have two location. Each
Does this number (you are dialing) has been ported from a different Telco?
When you dial from the other city and you get service not available you
may be dialing from a different Telco that either has no route aggreement
for the dialed network, or the number portability database (of Out of city
I got one of this boards and I got it successfully replaced by Avanzada7
(Digium official reseller) immediately.
On Nov 24, 2007 6:46 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Actually if you rule out all the clone tormenta cards (nothing wrong..
but very dated design... I wouldnt buy
Digium Cards have been just great on my experience and their support has
been simply the best one, via IAX (free Call) Remote Acess and hardware
config review and troubleshooting.
Many Thanks to Digium and their official reseller for Portugal and Spain
Avanzada7 great work!
Best regards,
Marco
with phone number in
the INVITE line whereas plugandtel put the callee number only inside the
To: Section.
Marco Mouta a écrit :
Could you describe in detail how did you fall into this situation, I
mean
the real example which SIP phone sends this invite? Is registered in
asterisk
Could you describe in detail how did you fall into this situation, I mean
the real example which SIP phone sends this invite? Is registered in
asterisk? it is a non-registered sip phone trying to dial a sip user at your
* box?
If this is an issue with a specific hardware outside of your asterisk,
${DIALSTATUS} will be one of:
- *CHANUNAVAIL* : Channel unavailable (for example in sip.conf, when
using qualify=, the SIP chan is unavailable)
- *BUSY* : Returned busy
- *NOANSWER* : No Answer (i.e SIP 480 or 604 response)
- *ANSWER* : Call was answered
- *CANCEL* : Call
as far as I know, softkey layout is managed by Cisco Call Manager and only
available running on skinny protocol.
On Nov 13, 2007 2:50 PM, Anciso, Roy [EMAIL PROTECTED] wrote:
There is an option to specify a softkey file in SEPmac.cnf.xml. I
have an email into our Cisco rep. I'm hoping he can
on wiki, just
wondering about php or something else
Best regards,
Marco Mouta
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that is possible data transmission with this Digium Card, I'm
wondering how... Any tip any tutorial?
Probably someone around the world as already done this before.
Best regards,
Marco Mouta
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hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
Sendmailhttp://sendmail.org/,
Postfix http://postfix.org/, Exim
Siemens GigaSet SL75
On 6/25/07, Michelle Dupuis [EMAIL PROTECTED] wrote:
We're looking at a large wifi phone deployment, and we're looking for
wifi phones that:
1. Are SIP compliant (Asterisk friendly)
2. Provision capable (ideally TFTP of own MAC address)
3. Industrial quality (no cheap
i believe www.voipango.de sell them to US
On 6/26/07, Nick Seraphin [EMAIL PROTECTED] wrote:
On Mon, 25 Jun 2007, Marcus Franke wrote:
Benny Amorsen schrieb:
MM Siemens GigaSet SL75
The SL75 is DECT, not Wifi.
Apart from that, was it really necessary to quote 20 lines and add a
and incoming
faxes.
Best regards,
Marco Mouta
On 6/22/07, Joe acquisto [EMAIL PROTECTED] wrote:
I have an existing Hylafax system using a mainpine 4 port board, 4 POTS
lines.
Have a recently installed Asterisk system, with a dedicated T1
line. (Well, it's really a fonality system).
What
pleease post your context exactly for the exten 5000 as u have it in live
system.
On 6/19/07, Douglas Garstang [EMAIL PROTECTED] wrote:
I have this in my dialplan…
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten = 5000,1,Answer
exten = 5000,n,Wait(1)
exten =
Siemens Gigaset SL75 are just Great!
On 6/12/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Tue, 12 Jun 2007, Deepak Naidu wrote:
We are planning to buy a wifi SIP phones to work with Asterisk 1.2-18
setup. I would like to get feedback views regarding Linksys WIP300
WIFI IP Phone or
so you r sure you have g729 licences installed and ur * is transcoding your
RTP streaming?
Test the work flow with disallow=all and allow=g729, can be my mistake but I
remember to read somewhere on the net any issue about codec negotiating
precedence when you use allow=all.
good luck
On
FYI,
http://www.voip-info.org/wiki/index.php?page=Asterisk+FAQ
*Can i install Asterisk on a beowulf cluster?* A cluster can't migrate
threads that use shared memory. Asterisk uses that kind of threads.So no,
Asterisk wouldn't work on a cluster. *(It might be helpful to know whether
anyone has a
backhole that would let
external users places PSTN calls through your server.
At the sametime if something goes wrong on outside world, your Lan VoIP
going will be kept 99,99% fully functional and let you make and receive
calls through PSTN.
Good Luck,
Marco Mouta
Ps. Qualquer coisa apita
Based on my experience I would say that using ${DIALSTATUS} variable would
be the most common way to do it...
On 4/23/07, Daniel Pittman [EMAIL PROTECTED] wrote:
G'day.
I am having reasonable success getting Asterisk 1.4.2 running and doing
what I want, but I can't figure out one particular
did you modprobe ztdummy?
On 3/30/07, Administrator TOOTAI [EMAIL PROTECTED] wrote:
Hi list,
we have a dual Xeon server with 2GB RAM running Debian Etch 2.6.18.4-686
kernel. The server has 2 B410P cards plugged in. No other card.
We installed Asterisk 1.4 trunk with zaptel trunk, ran make
Only with Asterisk you can handle it, but of course it depends on your
requirements on scalability and redundancy needed.
How many agents? How many diferent locations? SIP trunk to your telco or
PSTN ? Remote Agents at home?
Post more details on your requirements and I believe there are so
Hi,
This is a tool that allows you at any time and any place of your Dialplan
or Dialout Call file to dial a specific extension at a specific context,
even if you are not currently in the specific context.
example:
you are at [from-internal] context and you can say:
[from-internal]
exten=
take a look on Originate command for Asterisk manager interface to get web
page generating calls between the two boxes.
Easier I believe is to use SIPp to be used as an UAC that starts dialing to
your box1 and in the dialplan of this box make a dial for a Zap channel on
Box2.
You need to
check register expiration on polycom , probably is higher than 3600 sec
(default on asterisk) , so after this 3600 , imagine polycom as an expire of
6000sec, there's a gap of 2400sec that polycom isn't registred!
On 12/10/06, C F [EMAIL PROTECTED] wrote:
While what you say might/should help,
Try safe_asterisk , for an easy way to start asterisk in background, and
then connect with asterisk process running asterisk -rx
Now you can use exit, and by the way you may look on wiki diferent ways to
run asterisk.
On 1/23/07, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote:
Hi, all
Stupid
I don't know about SNOM, but with Xlite Softphone you can have the SoftPhone
internal dialplan.
Ex.
[29];match=1;pre=0; this adds a Zero to every nine digits number
s I dial begining with 2 or 9 , this has nothing to do with asterisk, is
VoiP phone dialplan.
So you can tell to the
enable rtp debug in your asterisk CLI and check if there's traffic passing.
Would be a first approach I think.
On 1/23/07, Tim Panton [EMAIL PROTECTED] wrote:
On 22 Jan 2007, at 07:28, Troy - Purple Oranges wrote:
I am at a loss, I can terminate and receive calls via any of my
providers
to understand Load average
results with Top command while incrementing calls dial from sipp to
asterisk, and how to determine max calls on Asterisk. This max calls is
defined when Sipp calls to * starts being discarded?
Best regards,
Marco Mouta
On 1/23/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote
this architecture, you
can setup as much IAXModem as your servers can handle, so it's very
scalable.
Best regards,
Marco Mouta
On 1/22/07, Ardjan Zwartjes [EMAIL PROTECTED] wrote:
Dear list,
The company I'm working for is trying to use app_rxfax to receive faxes on
an Asterisk machine. Our initial
Fax Server is not Asterisk, but some
one had done it already and it's widely used Hylafax...
Please let me know if i'm missing something on this email.
Best regards to this great Community,
Marco Mouta
dCAP
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Take a look on:
Dialplan applications:
GetGroupMatchCount([EMAIL PROTECTED])
SetGroup([EMAIL PROTECTED])
Using this two applications you can deploy a max calls control inside your
dialplan!
check this too: http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup
Hope it helps
On 1/19/07,
perfect.
But in my case i didn't try that. If someone has a SPA942 on their own lab
and can try this without damaging the phone would be nice info to share, I
believe!
Best regards,
Marco Mouta
On 1/17/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Andrew Joakimsen wrote:
I too seem
Freepbx GUI let's you create different administrators with different
permissions!
On 1/17/07, Kate Kretz [EMAIL PROTECTED] wrote:
I like the idea of Virtual PBX, but I don't like python language.
Are there other implementations ?
I'd like some java or php thing.
On 1/16/07, Tzafrir Cohen
My mistake Tzafrir, you are right!
On 1/17/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Jan 17, 2007 at 11:47:35AM +, Marco Mouta wrote:
Freepbx GUI let's you create different administrators with different
permissions!
But can you separate the permissions by context/domain
dialplan.
That is where I would start.
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Marco Mouta [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi all,
I've an asterisk 1.2.5 running very well for about a 9 months, and
suddenly i cannot dial extensions 4XXX from SIP Phones.
Now comes
You may use astdb for this.
Just set an entry on AstDB with user password and then for every outgoing
call prompt an audio to introduce password and then check if it exists on
AstDB.
User may be the caller ID and the pass is introduced by DTMF.
Then you may use a GOTOIF to allow or not
point me out where is the problem! This server has only
sip extensions.
P4 - 1G RAM wiht TE110P with weekly reboot.
Best regards,
Marco Mouta
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4XXX numbers that exist on my server nothing happens and i get
call failed: Request timeout.
Calls from PSTN to this SIP extensions 4XXX work FINE.
The context is fine, this was working for long time. suddenly seems to get
broken.
Hope someone can help me on this.
Best regards,
Marco Mouta
post here your extensions.conf
On 1/8/07, Mattias Andersson [EMAIL PROTECTED] wrote:
Hi!
Unfortunately did this stop Asterisk to register ny phones and trunk.
Did I put tit in the wrong place?
//Mattias
Hi!
Exactly what I needed.
It was the 209 part that I did not figure put.
Thanks!
That's what i told you Mattias.
On 1/5/07, Mattias Andersson [EMAIL PROTECTED] wrote:
Hi!
Exactly what I needed.
It was the 209 part that I did not figure put.
Thanks!
//Mattias
At 03:53 2007-01-05, you wrote:
exten = _9070X./209,1,NoOP,SORRY CHARLIE
exten = _9070X./209,2,Congestion
This
Hi all,
I was having a similar issue, using TE110P from Digium all incoming faxes
were detected and correctly received.
When trying to send outbound faxes, they all get broken... I do believe it
may be related with Echo Cancel enabled on my Zapata.conf, any ways i've set
also fax detect for
Hi Joao,
I'm not very experienced with SNOM, but have you though about providing fix
IP for you VoIP hardphones?
That way you could avoid the registration problem. At least while you don't
get your final solution.
Hope it helps,
MoutaPT
On 1/3/07, Joao Pereira [EMAIL PROTECTED] wrote:
Hi Mattias, add this to your dialplan:
exten= _/CALLERIDNUMBER,1,Hangup()
; Basically you are doing a pattern match with callerid match on your first
priority!
; You may keep your remaining dialplan, no changes needed
Pls Give me some feedback
Best Regards,
MoutaPT
On 1/3/07, Mattias
Are you sure there are no VoIP Phone users with Eyebeam or even polycom
requesting SUBSCRIBE for other extensions?
It happened to me, that users on my network were adding Subscribe for PSTN
numbers that aren't even extensions on my * server.
On 12/29/06, Lorentz Hinrichsen [EMAIL PROTECTED]
Does the user who is running asterisk has permissions to execute it? check
you script file permissions.
On 12/22/06, Andre Gustavo Lomonaco [EMAIL PROTECTED] wrote:
Hi,
I created a script named example2.sh which goal is read some text from my
HP Service Desk using an application in java and
,
Marco Mouta
On 12/15/06, John French [EMAIL PROTECTED] wrote:
I'm simply trying to forward calls to users who have the call forwarding
feature enabled (FWD Status and FWD Ph Number kept in the astDB). The
problem is that I want users to be able to forward calls to numbers that
they would normally
forking CDR could help Ricardo.
On 12/15/06, Ricardo Martins [EMAIL PROTECTED] wrote:
Hi John, I´m very interested into this call forwarding capabilities and
I solved this problem filtering on the web-script (in my case, php) the
number the user can intert on the database. (I know it´s not an
number.
After this Hands on I can sucessfully send faxes with Hy-email2fax --
Hylafax---asterisk Sucessfully.
But as i mentioned before i need to get ride of ^M on the subject line.
Any one can help me on this?
Best regards,
Marco Mouta
On 12/13/06, Lee Howard [EMAIL PROTECTED] wrote:
Marco
/etc/asterisk/modules.conf
On 12/13/06, Angel Heart [EMAIL PROTECTED] wrote:
Hi,
In what Asterisk file can I edit for me to temporarily unload such
modules. But later I woudl still be able to load them.
Thanks
Angel
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me too, i'm trying to add sip users , i click save, it reports successfully
saved... but there are no sip accounts created...
On 11/29/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote:
i had the same problem. the GUI stopped responding to configuration
changes.
On 11/28/06, James Willing [EMAIL
take a look on Audacity program is opensource and has the option Generate
Beep, then just add some Gain as you want...
On 12/2/06, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
We've had a few people complain that the beep before leaving a
voicemail is not loud enough and too short. Does
or update options visit:
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the call to phone_B. A REFER message is than routed backwards
to Asterisk, and he replies with those 404 Not Found messages. Phone_B
never gets called!
Should Asterisk be registered in Ser so this can work properly? How can
that be done?
Thanks,
Ricardo.
Marco Mouta wrote:
Hi Ricardo
,
Marco Mouta
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a scroll on this to display everything? do i need to
resize the buttons?
For sure someone now how to solve this basic question:)
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://lists.digium.com/mailman/listinfo/asterisk-users
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Marco Mouta
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Hi all,I've 2 * servers with static IP, and i notice that if i set both sip peers with host=server_ip and qualify=yes it presents UNREACHABLE on asterisk CLI.When i changed the host parameter to host=dynamic and set the register string in the [general] of
sip.conf on both servers, the connection
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-- Com os melhores cumprimentos,Marco Mouta
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callwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0group=1callgroup=1
pickupgroup=1immediate=noThanks Marco-- Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br2006/11/9, Marco Mouta
[EMAIL PROTECTED
Hi guys,
I've been looking on wiki, but i could find it only for chan_capi:
http://www.voip-info.org/wiki/view/Asterisk+PBX+functions
In the CAPI channel
See Asterisk CAPI channels
* Call Deflection (CD) (redirect without answering): Implemented
by chan_capi
How can i do it with my
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jb
Marco Mouta a écrit :
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face
specified
and no allow and/or deny restrictions at all. If such an entry is
found, accept the connection. and use the name of the found iax.conf
entry as the connecting username.
Pls give some feedback if you solved the problem.
On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi
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?
Mark
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My mistake:
[kpn-is]
exten= _X.,1,answer
exten= _X.,2,Noop(My telco is sending me this MSN string: ${EXTEN})
exten= _X.,3,wait(1)
exten= _X.,4,Playback(vm-goodbye)
exten= _X.,5,hangup
On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:
Plse Read bellow:
On 10/27/06, Mark Hannessen [EMAIL
...
jb
Marco Mouta a écrit :
Hi,
I think i found your problem, look that in your debug you have, -
Accepting UNAUTHENTICATED call from 10.0.0.160:
Take a look on incoming call authentication, and how asterisk handles this:
http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication
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