+= ../staging/echo/
with
obj-m += ../staging/echo/echo.o
With the original one I got some warning messages about oslec symbols
not defined. I think that the builder was not able to find the oslec
object file.
Am I doing something wrong?
Thank you and best regards.
Marco Signorini
numbers using the
dialplan ??
Am I too far here ? or is this something that already exists
and I don't know it ??
I would appreciate any help.
Thanks a lot,
__
Marco Eduardo Cordeiro
Visioncom IT
Visioncom Tecnologia da Informacao
Asterisk 1.4.20.1 built by root @ Gateway on
a i686.
Is this the correct behavior or a bug?
Thank you and best regards.
Marco Signorini.
Steve Gladden wrote:
Scratching my head and trying this.
Asterisk Version: Asterisk 1.4.21.2
Tried:
exten = 4771,1,ExtenSpy([EMAIL PROTECTED])
exten
with a 100
TRYING message.. but it never send an ACK. At the same time, the
t38modem is producing the log I've attached below (sorry for the long
post).
Any help is appreciated.
Thank you.
Marco Signorini
2008/09/22 23:53:39.395 Opal Liste...er:80b95c8 SIP PDU Received on
udp$192.168.0.5:5060if=udp
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
applications everyday.
I have made some stress call tests, using all available CICs at once, and
had no problem at all.
Congrats to the perfect development of the SS7 support to Mr. Fredrickson.
Hopefully soon we'll have MAP support as well.
Marco
-Mensagem original-
De: [EMAIL
Hello,
Just wanted to let you know that the XP version works fine on vista.
I was working on a similar program but didnt have enough time to finish, I
was working on Delphi 7 btw.
Thanks
Marco.
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Gerald
You have to limit calls to these agents, use incomminglimit or call-limit on
sip.conf to do that.
That way, when the first agent answers a call, all the other calls directed
to it will return with busy signal, and will be transferred to the other
agent.
__
Marco
?test,s,1)
Rgs,
Marco Cordeiro
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Nhadie
Enviada em: quarta-feira, 30 de julho de 2008 16:47
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: [asterisk-users] GotoIftime
Hi
How cn i define
Have you tried incominglimit=1 on sip.conf ??
It worked for me, no matter which softphone or ipphone / ATA I use, it
works.
You have to use it inside the configuration for every sip peer, just like
this:
[1002]
Type=friend
Host = dynamic
Port = 5060
incominglimit=1
.
.
.
De: [EMAIL
call-waiting
Hello
thank u for ur attention but I did it and in fact its the same as call-limit
in newer versions.
this cmd limit ur call not disable call-waiting.
best regards
On Wed, Jul 23, 2008 at 5:02 PM, Marco Eduardo Cordeiro
[EMAIL PROTECTED] wrote:
Have you tried incominglimit=1
some reasonable price ethernet DIN rail
industrial controls that provides HTTP capabilities and you can write a
simple AGI script that generates some HTTP transactions to set the board
rele' status to whatever you want.
Best regards,
Marco Signorini.
c james wrote:
I have an opportunity to interface
faxdetect=no
channel = 1
I'm using zaptel-1.4.6 and asterisk-1.4.20.1.
I hope this could help you.
Best regards,
Marco Signorini.
Enrico Maistro wrote:
Hi,
I'm trying to get up and running a TDM400 with a standard italian pots
line but i'm having
problems at getting asterisk
be that the problem is related to Asterisk 1.6? Unfortunately I
never had the possibility to try this new version.
Best regards,
Marco Signorini.
Enrico Maistro wrote:
My zapata.conf differs in:
language = it instead of en
rxgain = 0.0 instead of 3.0
jbenable = no instead of yes
Unfortunatly even
for the correction.
Best regards,
Marco Signorini.
Tzafrir Cohen wrote:
On Mon, Jul 14, 2008 at 08:56:40PM +0200, Enrico Maistro wrote:
Hi,
I'm trying to get up and running a TDM400 with a standard italian pots
line but i'm having
problems at getting asterisk to detect when the line get answered
as DTMF Caller ID type, but still not working.
Let us know what kind of problem you have, maybe I can help you out.
Marco
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Doug
Enviada em: quarta-feira, 18 de junho de 2008 15:57
Para: Asterisk Users Mailing
Personally, I love the debian way, but I must admit that when it gets
to Asterisk, I prefer to use a RedHat-based distro like CentOS, first of
all for the proven reliability, then for the widely used rpm packaging
system and last because there are many distro CentOS-based that provide
a stable
Alan Lord wrote:
If you only have one analogue line why not just get a simple x100p card?
When you use OSLEC with them they work great here in the UK. I bought my
card from a USA based eBay seller. Total cost for card and shipping was
about £17.00
Respectfully, I don't agree. I've
Lately I've been offering these stuff to a customer; a valid solution is
the one provided by Jabra with their DECT headset (see
http://www.jabra.com/Sites/Jabra/UK-UK/products/Pages/JabraGN9330.aspx )
and a electronic lifter as the one for the Snom phones here (
do you think?
Bye
Marco
Alan Lord ha scritto:
Hi there,
in case anyone is interested, I've just taken ownership of a small home
network (3 handsets) of the brand new Siemens DECT PSTN/VOIP phone.
It works great with Asterisk. Here's my overview and review so far...
http
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
matter is that I have NO clue on where to append this code for
outgoing calls from these specific extensions.
If anyone has a simpler idea (yeah, mine is pretty twisted %) ) or knows
how to put a code before the dial string of an extension, let me know!
Thanks in advance,
Marco
That makes PERFECT sense and also makes me aware that I need to review
asterisk theory :-P
I'll put it under test and let you know how it works.
Thanks a lot!
Marco
Rodrigo Gonzalez ha scritto:
Create different contexts and assign them to the extensions
[trunk1]
exten = .X,1,Dial
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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what i need is really near to chan_mobile but i don't need
Bluetooth connection...only USB.
Thanks again
Marco M.
p.s. am i writing in the right place or should i write to another
asterisk-related mailing-list?
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to try this wireless solution in order
to be reached by a phone call i.e. when i'm on a train using my laptop
(where i would have asterisk running).
Do i need a particular driver to do so?
How can i make asterisk look at my usb port?
Thanks a lot
Marco Maso
Thanks Michael,
that's a *huge* thing you're telling me, in the wiki details for the
PCI-X bus I've read about retrocompatibility, but I just wanted to be
100% sure. I can go on and order my server, now!
Thanks again
Marco
ps. This proves also the complete unaccuracy of the information
,
Marco
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for Outage during one month is 0,432 minutes
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.
Best regards,
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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confidencial para uso exclusivo do destinatário. Se não for o
, 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
bad english.
--
Marco Carvalho (macs) | marcoacarvalho(a)gmail.com
Maceio - Alagoas - Brazil
Debian GNU/Linux AMD64 unstable (Sid)
GNU-PG ID:08D82127 - Linux Registered User #141545
Notícias Semanais do Debian em Português: http://www.debian.org/News/weekly
Alertas de Segurança Debian (DSA): http
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 9/27/07, Jonn R Taylor [EMAIL PROTECTED] wrote:
marco britannio wrote:
Hi all,
I'm trying to setup an asterisk based fax receiving machine.
i'm using asterisk 1.2.18 and app_rxfax with spandsp 0.0.4pre9
I have no problems with a modem-fax, but with the fax machines i have
tried
receive not successful - result
(11) Unexpected message received.
Sep 26 17:26:18 DEBUG[4741] app_rxfax.c:
==
can anybody help me?
thank you in advance,
marco
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C F wrote:
BTW, AFAIK, there is no such thing as host=static it's either dynamic
or an IP/Name.
Yeah, I learned that the hard way. I had only set up dynamic devices
for a couple of months, and the first time I had reason to set up a
device with a static IP, I just assumed that
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,
I've recently upgraded Asterisk to the latest version 1.4.9
on a PBX that manages several queues, but at least on one queue strategy
(leastrecent) it doesn't seem to be distributing the calls has it should.
I think this strategy should work like
basic bugs that I can't understand why
they still happen... I'm one of the very few persons that use it for queues?
Thank you for your comments and help.
Cheers,
Marco.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jakub Glazik
Sent: sexta-feira
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
the phone to talk, no
sounds/voice gets through between phones.
Any help would be appreciated !
Thanks,
Marco
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2 problems in order to find what's happening
?
Thanks
marco
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unuseful mails.
Regards
marco
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Dave Cotton
Inviato: martedì 15 maggio 2007 19.17
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] Trixbox problems
On Tue, 2007-05-15 at 19:57
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
and forum! :-)
Thank all, regards
--
Marco Ciacci
Asterisk Admin
Windows Server Linux Admin
Security Networking
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, but i didn't find nothing :-(
Bye,
At 16.05 27/04/2007, you wrote:
On Fri, 27 Apr 2007, Marco Ciacci wrote:
HI all!
I'm looking for some infos to configure stun server support for a SIP peer.
I've installed Asterisk 1.4.3, but searching for stun support in
chan_sip (sip.conf) i've found nothing
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=
with the usual command ztcfg and it is
strictly related to the presence of the echo canceller onboard.
Thanks a lot
Marco
Looking for earth-friendly autos?
Browse Top Cars by Green Rating at Yahoo! Autos' Green
Hi Michelle,
actually, I didn't try it...
The server is a HP Proliant ML150T G3.
Currently I'm not in the condition to follow your suggestion, but I hope in the
near future to be able to give you a feedback.
Thanks!
Marco
Have you tried starting Linux with irqpoll / noapic? Sounds like
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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To UNSUBSCRIBE or update
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]
Hi, All
How do I install Zaptel drivers on a system running Suse?
Make results:
grep: /lib/modules/2.6.16.13-4-smp/build/include/linux/autoconf.h: No existe
el fichero o el directorio
make[1]: Entering directory /usr/src/asterisk/zaptel/zaptel-1.4.0-beta2
make -C
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
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