We're looking at using a number of Dell Poweredge SC430 servers as Asterisk hosts in our smaller overseas offices with Digium cards in to provide local breakout over the pre-existing analogue or digital phone lines (One office uses ISDN2 the others analogue)I note that the SC420 is listed as incomp
Jean-Michel Hiver wrote:
3) The G723 codec also does VAD (which Asterisk doesn't support).
Shame it doesn't... if you could do IAX2 trunking with g723 5.1kbps +
VAD, that'd be awesome for narrow links (which is very common in
developing countries).
On Tue, Sep 05, 2006 at 09:56:57PM +0200, Olivier Saulnier wrote:
> Hello,
>
> I would like for some reasons, catch the ring event since Asterisk, in
> real-time. Is this information record in a database? How can I read it,
> immediatly?
> I either think to catch the information by a little shel
What signaling method i should use for connecting Asterisk(Gentoo,
Tormenta 2) + Samsung OffServ 500 by PRI flow?
What parametrs in zaptel.conf, zapata.conf?
---
Какой метод сигнализации нужно использовать при подключении
Asterisk(Gentoo, Tormenta 2) + Samsung OffServ 500 по PRI потоку?
Интерес
Thanks for the corrections.
What would I call a unit that accepts a PRI connection with a 10-channel PRI and
an Ethernet output?
I am looking into the Tellabs
echo canceller now. Digium’s echo canceller offering is not suitable for
my 1 PRI project because it is probative cost wise.
Hello all,
Another question related to Realtime.
Is it possible to call macros using Realtime arch?
I have a macro definition in table extensions_conf in my MySql db as:
30 | macro-stdpbx1exten | s |
1 | SET| fwdedNum=${DB(CFWD/${ARG1})
So, I don't understand the reason to use ## then. I think you are wrong.On 9/6/06, David Gagnon <[EMAIL PROTECTED]
> wrote:
No, when you press the
first time, Asterisk is in standby waiting for the other one.
David
De :
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] De
Xue Liangliang wrote:
Hi, all, I am not sure whether we can install both TDM400P and TE110P in
the same machine, sometimes our customers have this kind requirement.
And further more, is it possible to install both TE110P and TE410P in
the same machine?
Regards,
Liangliang
it all depends on
I was able to track this problem down and solved it with a little
manual reading.
I used the remote access features documented on page 3-189 in the
Revision 7 manual with a few security tweaks.
I will soon update the wiki with my procedures.
Thanks guys.
Sterling.
On Sep 5, 2006, at 6:00
Hi,
I was wondering if those more familiar with PSTN->Asterisk installs
could take a listen to this (86k recording):
http://confluence.atlassian.com/download/attachments/185668/linegoingbad.mp3?version=1
It's what I hear dialling into our Asterisk box. As soon as the call
receiver makes a sound
No, when you press the
first time, Asterisk is in standby waiting for the other one.
David
De :
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Michael Strelnikov
Envoyé : 5 septembre 2006
02:37
À : Asterisk
Users Mailing List - Non-Commercial Discussion
Objet : R
Look at queuemetrics.
Pretty good.
David
De :
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Technical Support
Envoyé : 4 septembre
2006 16:59
À :
asterisk-users@lists.digium.com
Objet : [asterisk-users] Call
center reports
Can someone point me to call ce
Hello all,Is there any way to know that call is finished? I know there are special tones sent by phone companies but how can I detect them and then configure Asterisk to use it?
Thanks,Michael
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Hi, all, I am not sure whether we can install both TDM400P and TE110P in
the same machine, sometimes our customers have this kind requirement.
And further more, is it possible to install both TE110P and TE410P in
the same machine?
Regards,
Liangliang
_
I need somebody who can test with me video phone settings.
I use Eyebeam!
Please contact me via MSN first: [EMAIL PROTECTED]
bye
Ronald
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No, I'm not looking for a voice talent.
I have been deploying an IVR in my company's China office, and our people
there complained about the way asterisk spoke the date in Chinese.
After discussing it with them, I have submitted a patch, which can be
found on the Digium Issue Tracker at
http://bu
I took a different approach with a Definity system and put asterisk
between the Definity and the telco. For my setup I had to create a
trunk group and a route then tell the Definity to pattern match and use
that new route, it is called ARS Analysis or something obscure.
In your setup, you wou
Well, currently I do not use them, but I hate to give it up either -
nice to be able to ID the call type and it might be useful some day.
Now it's a pain to deal with. What you get is the same here from GBX,
but with MCI, it the same with ANI II added to ANI
Bart
Steve Totaro wrote:
Do you us
Do you use the AnI II digits for anything? If not, call the telco and
tell them to just send ten digits. When I used to have some T1s with
UCN, they sent the ANI II digits in a separate field specifically for
that. I could see them in PRI debug on the console. Now with Global
Crossing, I ge
I was hoping someone might have the answer to this:
As an update: even though you can modify the Caller ID in
extensions.conf for call handling and use CallerID(number) in you
script, asterisk does not honor the modified number in CDR and
VoiceMail.- I need to fix the number at answer.
Bart
Hello List,
I was able to track down my issues with the PRI from a few weeks
back. Turns out the legend switch had a Terminal Equipment ID of 2
and was supposed to be a 0. Go figure. This was keeping the D-Channel
from coming up although the rest of the B-Channels were up. It was
very ea
To expand on what Eric said.. People commonly use _X. for what you're wanting.
It's just as effective, but doesn't match the "special" extensions.
- Eric \"ManxPower\" Wieling <[EMAIL PROTECTED]> wrote:
> Your problem is caused by using exten => _. DON'T DO THAT!
>
> When Hangup() is bein
(I'm getting 404 Not Found from the search engines)
I have a system that gets a deadlock every week or so. On the logs I
have many "channel.c:787 channel_find_locked avoided deadlock for
0x837730" messages.
The system has an Eschelon T1 with 6 voice (with dchan) arriving on a
TE110P. Aster
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Steven Ringwald wrote:
> Mike wrote:
>> Hi all,
>>
>> I just found out how to set the column userfield, in the CDR DB to
>> whatever I desired. Can I add multiple custom columns to the DB and
>> fill them from the dialplan, or is it limited to one c
Hi,
I need to send a message to an agent when the ACD starts to ring on he/she.
I have and application already built that sends such a message (just
like a cti), just don't know how to get from asterisk which agent was
selected prior to ringing him (or during ringing), so that I can get
in
Curt Shaffer wrote:
Here is my extensions_custom.conf. The WakeUp context will not work.
If I change the context name to say, CRAP, it works like a charm. Can
anyone explain this?
And the output from the console when you dial 611 and it doesn't work?
Doug
-- Ben Franklin quote: "Thos
Your problem is caused by using exten => _. DON'T DO THAT!
When Hangup() is being run then Asterisk will jump to exten => h Since
_. will match "h" it will go there.
Marco Mouta wrote:
Hi all,
I think i'm missing something very very basic! I want my calls with DID
48XX
(From pstn E1 TE1
Mike wrote:
Hi all,
I just found out how to set the column userfield, in the CDR DB to
whatever I desired. Can I add multiple custom columns to the DB and
fill them from the dialplan, or is it limited to one column?
I am using Asterisk 1.2.4 and MYSQL for the CDR DB.
As far as I know, i
Hi
all,
I got the following
"warning" in the console (using 1.2.4):
Sep 5 16:24:02 WARNING[4375]:
app_addon_sql_mysql.c:318 aMYSQL_fetch: ast_MYSQL_fetch:
numFields=3
Im not sure why I am
being "warned" that there are 3 fields returned by my query (It's what's
supposed to happen, th
Hi
all,
I just found out how
to set the column userfield, in the CDR DB to whatever I
desired. Can I add multiple custom columns to the DB and fill them from
the dialplan, or is it limited to one column?
I am using Asterisk
1.2.4 and MYSQL for the CDR DB.
Mike
___
Hello,
I linked an Asterisk server to a Brazilian PBX (Leucotron) through an E1
connection, using MFC/R2, that's common down here. The connection works
properly. I'm able both to dial and receive calls through that link,
among their extensions.
The problem is that the PBX configuration is very t
No, T.38 doesn't work with Asterisk. Only works with Asterisk
t38passthrough patch that you can find at URL:
http://bugs.digium.com/file_download.php?file_id=9335&type=bug
For me it only worked well with patch for version 1.2.4 of Asterisk.
Regards,
Ricardo.
Kokfoo Soo wrote:
Is T.38 fax
Hello,
The service I am using requires authentication.
In sip.conf, setting:
[authentication]
auth=name:[EMAIL PROTECTED]
Gets the authentication working for the INVITES but when I try a transfer, I
can see the REFER but then asterisk quickly says BYE. The provider sends back a
401 UNAUTHORIZE
[ Context 'from-internal-custom' created by 'pbx_config' ]
'1234' => 1. Playback(demo-congrats)
[pbx_config]
2. Hangup()
[pbx_config]
'h' =>1. Hangup()
[pbx_config]
Include =>'NewsClips'
[pbx_config]
Include =>'WakeUp'
[pbx_config]
--
Curt Shaffer wrote:
Here is my extensions_custom.conf. The WakeUp context will not work.
If I change the context name to say, CRAP, it works like a charm. Can
anyone explain this?
What does show dialplan from-internal-custom display?
Doug
--
Ben Franklin quote:
"Those who would give
Hello,
Has anybody configured Asterisk and the Articulation palm client to work ?
I can make calls but I cannot make it register to receive calls.
It does not register to the box. There are so few parameters that I
think Asterisk sip.conf must be changed somewhat.
I do not pass any parameters her
Hello,
I would like for some reasons, catch the ring event since Asterisk, in
real-time. Is this information record in a database? How can I read it,
immediatly?
I either think to catch the information by a little shell script as:
asterisk -r |tail -1|grep ring|awk ... and redirect the intern
Is T.38 fax work through Asterisk? I have the config below in my sip.conf, but the fax doesn't work and give me the CLI lines below. My current version is 1.2.10. Please help.[Inboundtopbx]type=friendcontext=pbxhost=10.18.188.84insecure=portdtmfmode=rfc2833canreinvite=nodisallow=allallow=g729allow=
Here is my extensions_custom.conf. The WakeUp context will
not work. If I change the context name to say, CRAP, it works like a charm. Can
anyone explain this?
[from-internal-custom]
exten => 1234,1,Playback(demo-congrats) ;
extensions can dial 1234
exten => 1234,2,Hangup()
e
Glad to help.
Happy dialling.
On September 2, 2006 23:05, Nick Ellson wrote:
> Hi Tim,
>
> The dial plan trick worked great. Added |40[01]x| to my plan and 4000-4019
> connect instantly from the PAP2 :) Added it to my X-Lite as well, and
> worked there too.
>
> Thanks!
--
Tim St. Pierre
IP tel
Does anyone know what the options are for the meet-me recording formats?I can't seem to find any documentation on the ${MEETME_RECORDINGFORMAT} variable and what it can be. Michael Lively[EMAIL PROTECTED]229-316-0011 ___
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I have not experimented with it lately but I think that is how it is
supposed to work.
The speedial buttons can be programmed to do BLF+speedial to a given
extension. If your getting an incoming call from one of the speeddial
extension as indicated by the BLF status you do not pick it up by press
equis software wrote:
Could I use different music on hold between waiting calls in queue and
calls that are waiting to be tranfered?
Yes, with the SetMusicOnHold command.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetMusicOnHold
Doug
--
Ben Franklin quote:
"Those who would
Hi
I am tyring to connect two * boxes over IAX with rsa, but I am having a slight
problem. It just doesn't work. My configuration looks like this:
iax.conf on box 1
[asterisk2]
type=friend
context=main
auth=rsa
inkey=asterisk2.mydomain.com
outkey=asterisk1.mydomain.com
host=asterisk2.mydomain.
Could I use different music on hold between waiting calls in queue and calls that are waiting to be tranfered?Thanks
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Lenny wrote:
Hello,
What are some solutions folks are using for faxes (inbound)? I was
considering the Stanafax option.
**Regards,**
**---***
**LB***
I got an 500 Internal Server Error when I entered my email address.
Thanks,
Steve
_
You should probably ask this in the VICIDIAL forums:
http://www.eflo.net/VICIDIALforum
Your problem is a known bug in the 2.0.1b1 release that has been fixed
in SVN 2-X trunk.
MATT---
On 9/5/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Hello all!,
Ive install all package of
Timothy R. McKee wrote:
I just ran an SVN update to attempt resolution of this issue and now there
is a completely different issue...Very strange.
1. inbound call comes into phone A and is answered.
2. transfer button pressed
3. number of phone B is entered
4. phone B rings and is answere
Michael Araba wrote:
I believe a hardware solution might be the way to go. Does anyone have a
suggestion of a cheap used or refurbished echo canceller I could use?
http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers
Doug
--
Ben Franklin quote:
"Those who would give up Ess
I was hoping to find a add-on package for my current configuration. I don't
need and completely new platform. I have looked at the two previous posted
websites. Scopserv is a total package solution from what I gather and iotum is
still in beta.
Thanks for the quick replys!
-Original M
Hello,
What are some solutions folks are using for faxes
(inbound)? I was considering the Stanafax option.
Regards,
---
LB
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asterisk-user
You could implement this very easily yourself.
Just write a small webpage that saves the user's find-me/follow-me
extension to a text file somewhere (or a database of course)
Then write a small agi, that checks for the file (or db value) and sets
a variable to jump to that extension
M
>> Has anyo
On Sep 5, 2006, at 11:56 AM, Michael Araba wrote:
I have had a bad experience with Asterisk and a Carrier's channel
bank.
The carrier brought in a PRI (data/voice integrated), the data and
voice
channels are split from the channel bank. I connected Asterisk to the
channel bank via T1 cros
Hi Roger ,
Has anyone developed a web interface where users could setup their own
find-me/follow-me services?
Yes, this is available on the ScopServ Telephony GUI (Commercial).
--
Joel Vandal, CTO
ScopServ Inc.
http://www.scopserv.com.
___
--Ba
Hello all!,
Ive install all package of Vicidial and astguiclient as I read
on a scratch install notes in a CentOs 4 with trixbox. But when I use some AGIS
added in a dialplan of the install documentation i get some sintax error on this
scripts like agi-VDAcloser_inboundCIDlookup.ag
Please post your misdn-init.conf as well as misdn.conf so i can try to help uOn 9/5/06, Giorgio Incantalupo
<[EMAIL PROTECTED]> wrote:
Hi,I hava an Asterisk box with a monoBRI + install-misdn-mqueue 0.3.1-rc23package installed.I can make outbound calls but cannot receive any. I get no Asteriskmess
I just ran an SVN update to attempt resolution of this issue and now there
is a completely different issue...Very strange.
1. inbound call comes into phone A and is answered.
2. transfer button pressed
3. number of phone B is entered
4. phone B rings and is answered. audio between A and B
Has anyone developed a web interface where users could setup their own
find-me/follow-me services?
Roger Workman
Business Development
Upperclassman/Universal Holdings LLC
Voice: 304.324.3800
Fax: 304.324.3801
ICQ: 4447584
FWD Network: 56505
Website: http://www.upperclassman.net
Billing Questio
I use rtcachefriends=yes and any changes I make in my database become
effective immediately along with also getting the MWI functionality.
Even though what you say makes sense. Go figure!
Ben, yeah if it shows it's loaded then it's there for sure. Sorry I
asked for it as in your module listing th
At 03:21 AM 9/5/2006, you wrote:
exten => 1,n,set(INNAT=${FIELDQTY(+,${ATELNO})})
None of them work.
This is what I do:
exten => s,n,gotoif($["${EXTEN:0:2}" = "+1"]?fixcid:okcid)
exten => s,n(fixcid), set(xxx=${EXTEN:0:2})
exten => s,n(okcid), noop()
Ira
___
I have had a bad experience with Asterisk and a Carrier's channel bank.
The carrier brought in a PRI (data/voice integrated), the data and voice
channels are split from the channel bank. I connected Asterisk to the
channel bank via T1 cross cable with a Digium T205.
On many calls users hear thems
Marco Mouta wrote:
Hi all,
Do you think it could be an affordable solution using a two fxs ATA
device to connect an old legacy pbx (with few users) with a main
asterisk server.
phonesanalogueSmallOfficeLegacyPBxATA-2FXS-SIP--MainOffice
AsteriskServer
This way also I w
Hi all,Do you think it could be an affordable solution using a two fxs ATA device to connect an old legacy pbx (with few users) with a main asterisk server.phonesanalogueSmallOfficeLegacyPBxATA-2FXS-SIP--MainOffice AsteriskServer
This way also I would use ATA device as a Trunk w
Hi,
I hava an Asterisk box with a monoBRI + install-misdn-mqueue 0.3.1-rc23
package installed.
I can make outbound calls but cannot receive any. I get no Asterisk
messages on the console except for these:
P[ 1] GOT IGNORE SETUP
P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE]
P[ 1] release_chan: Ch not
Evenin' (o;
Following strange problem:
7970G SIP phone <-> asterisk <-> SIP provider
In sip.conf I register to my SIP provider to receive
calls from them...but as soon the numer rings I
see as CallerID the configured outbound number
from my SIP account and not who is actually calling...
So
Can I configure different MOH for waiting calls than parked calls?Thanks
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Thanks Peter, I've also learned with your tips ;)On 9/5/06, Peter Bowyer <[EMAIL PROTECTED]> wrote:
On 05/09/06, Jay Moore <[EMAIL PROTECTED]> wrote:
> Perhaps if answering the simple things politely is too difficult for> you, you'd be better off not answering at all. Someday, I hope, you'll> find
> -Ursprüngliche Nachricht-
> Von: Koopmann, Jan-Peter [mailto:[EMAIL PROTECTED]
> Gesendet: Dienstag, 5. September 2006 13:54
> An: Asterisk Users Mailing List - Non-Commercial Discussion
> Betreff: [asterisk-users] Experience Patton BRI gateways and Asterisk?
>
> Hi,
>
> can anybody com
HI,I have successfully integrated CallManager and Asterisk and was able to make call from one of Asterisk phone to CallManager Phone.But Could not able to make call from CallManager to asterisk.I have also tried the below link :- http://www.voip-info.org/wiki/index.php?page=Asterisk+Cisco+CallMana
I wrote a little patch to app_queue.c so that the function QUEUEAGENTCOUNT
will only return members that are not busy.
My dialplan goes something like this (in AEL):
SET(QACFREE=${QUEUEAGENTCOUNT(abcstaff|free)});
if (${QACFREE} > 0) Queue(abc|trnd|||20);
So if th
If you want to use MWI, and I imagine most people would, you have to cache your
realtime data, which means that changes to the tables do not become effective
immediately. They become effective after you prune the entry in memory.
Doug.
> -Original Message-
> From: RR [mailto:[EMAIL PROT
Sorry folks.. all is well and the options are now being triggered..
The problem was that while I was configuring the settings I didn't fill in
the mode from working on this last week :)
Silly mistake; but its all up and running ..
I'm sure I'll be back soon, but until thank take care and thanks!
Ok ..
So I have moved asterisk to an unrestrictive line and still once the IVR
gets going; any keys pressed don't trigger any of my menu options.
I have tried all sorts of settings in the sip.conf.
John, you mention to switch modes in the trunks/sip.conf, but how can I tell
the provider Stanapho
On 9/5/06, Jean-Michel Hiver <[EMAIL PROTECTED]> wrote:
Have you tried progressinband=yes? As far as understand it, it forwards
early RTP (that is, stuff that is received prior to the ANSWER), which
might just do the trick.
Hm, I have just added this in zapata.conf and sip.conf, and also tried
Virmones Pereira Tavares de Miranda schrieb:
How to configure asterisk and zaptel for ISDN ? EWSD?
Hi,
below is the ISDN part of my zaptel.conf.
Imho crc4 is software selectable in EWSD, thus
ask your provider! The D-channel could be found
at another location, thus ask your provider!
For T1
thank you for your help.
I'll try as soo as I can.
Oscar Bossi
>Messaggio originale
>Dal: [EMAIL PROTECTED]
>Data: 05/09/2006 15.55
>A: <[EMAIL PROTECTED]>, "Asterisk Users Mailing List - Non-Commercial
Discussion"
>Ogg: Re: [asterisk-users] LinkSys PAP2 ATA & Siemens Cordless 3010
>
>
Hi,
I have the following problem.
I need queue because of dynamic agents but I only want service as many
callers as available members are and want zero length waiting queue.
For example. I have two queues (q1,q2) and I use AddQueueMembers and
RemoveQueueMembers for maintain queue members.
exte
yeah...I Got a Siemens Phone and i can't hear the ringing.
Try to change these settings in the pap2 device (Admin -> Advanced Mode->Regional settings)
Voltage = 90V
frequency = 20 Hz
impedance = 900 ohms
waveform = trapezoidal not sure about the question but this is a must i th
stoffell a écrit :
On 9/1/06, Xue Liangliang <[EMAIL PROTECTED]> wrote:
Hi, all. I am from Singapore, we deployed a few PABX based on Asterisk.
Here in Singapore there are two Teleco providing E1 pri service, we
encountered a strange problem : when calling a number that is
unavailible or line
How to configure asterisk and zaptel
for ISDN – EWSD?
Its possible?
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Hi everyone,
I'm having a problem using this cordless with this ATA.
When I try to call that phone, the line is busy. When this phone tries to call
someone, no line up.
Ata is working with another phone that's not a cordless so it's configured
correctly.
Any clue about this problem?
Best reg
hello,
Since a few days I'm looking for the 'best' way to get the telco
"error" messages when dialing wrong/busy/non-existing numbers. I can't
get it to work on E1 or ISDN BRI.
An alternative option is to detect the hangup_cause (no problem here)
and play our own voice prompts. I would like to a
3) The G723 codec also does VAD (which Asterisk doesn't support).
Shame it doesn't... if you could do IAX2 trunking with g723 5.1kbps +
VAD, that'd be awesome for narrow links (which is very common in
developing countries).
___
--Bandwidth and Co
On 5 Sep 2006, at 13:21, Stefan Tichy wrote:
On Tue, Sep 05, 2006 at 11:21:43AM +0100, Tim Panton wrote:
exten => 1,n,set(INNAT=${REGEX("^\+","${ATELNO}")})
exten => 1,n,Set(PLUS=\\+)
exten => 1,n,set(INNAT=${REGEX("^${PLUS}" ${ATELNO})})
That worked, Thanks!
Tim.
Tim Panton
www.mexu
On Tue, Sep 05, 2006 at 11:21:43AM +0100, Tim Panton wrote:
> exten => 1,n,set(INNAT=${REGEX("^\+","${ATELNO}")})
exten => 1,n,Set(PLUS=\\+)
exten => 1,n,set(INNAT=${REGEX("^${PLUS}" ${ATELNO})})
If it is an extension, this should work too
exten => _+.,1,Goto(011${EXTEN:1},1)
--
Stefan Tichy
Roland wrote:
I've tried all those at voip.info.org but I just couldn't get it
right. and I don't have the luxury of time to try figure out how to
make it work by myself.
The official FreePBX install docs (which have Asterisk instructions as
well) for CentOS are here:
http://aussievoip.com/w
Hi,
can anybody comment on patton inalp voice gateways and Asterisk? How good is
there echo cancellation? How good is the interop with Asterisk? I am especially
looking for reports on 4630 and 45xx series with BRI.
Thanks a lot in advance!
Kind regards,
JP
__
www.nerdvittles.comOn 9/5/06, Roland <[EMAIL PROTECTED]> wrote:
I've tried all those at voip.info.org but I just couldn't get it
right. and I don't have the luxury of time to try figure out how tomake it work by myself.any other very useful new guides you guys have? tnx_
On 5 Sep 2006, at 12:04, Doug Lytle wrote:
Tim Panton wrote:
exten => 1,n,set(INNAT=${REGEX("^\+","${ATELNO}")})
Just a grab in the dark, have you tried using single quotes instead
of the double?
Sadly not, with :
exten => 1,n,set(INNAT=${REGEX('^\+','${ATELNO}')})
I get :
Sep 5 12:
I've tried all those at voip.info.org but I just couldn't get it
right. and I don't have the luxury of time to try figure out how to
make it work by myself.
any other very useful new guides you guys have? tnx
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Tim Panton wrote:
exten => 1,n,set(INNAT=${REGEX("^\+","${ATELNO}")})
Just a grab in the dark, have you tried using single quotes instead of
the double?
Doug
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Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty
Hi!
> >>> Does any of you knows an Hardphone with VPN client embedded?
> >> Take a look at Zultys SIP phones. VPN enabled.
> >>
> >> www.zultys.com
>
> As I too am interested in IPsec capable hardphones (or ATA's), do you have
> a suggestion what to look at instead?
>
> I mean: It's nice to say
I'm hoping someone has solved this problem before, because I'm stuck!
I get phone numbers from a database into my dial plan via AGI.
Some of the numbers use the + sign to denote an 'international'
number.
I need to re-write these numbers into a format my IAX provider
can deal with (ie a us style
Thank you Very MUCH I really appreciate your explanation, i wasn't getting it!On 9/5/06, Tony Mountifield <
[EMAIL PROTECTED]> wrote:In article <
[EMAIL PROTECTED]>,Marco Mouta <[EMAIL PROTECTED]> wrote:>> I've solved the problem, but still not understanding very well why do i need
> it:>> I've
EHLO (o;
Anyone succeeded with hooking up a Matra 6501 PBX to * ?
cheers
rick
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In article <[EMAIL PROTECTED]>,
Marco Mouta <[EMAIL PROTECTED]> wrote:
>
> I've solved the problem, but still not understanding very well why do i need
> it:
>
> I've inserted inside [ext-did-custom]
> exten=>h,1,hangup
>
> Why do i need this? this is not usually used to run something after an
>
On 9/1/06, Xue Liangliang <[EMAIL PROTECTED]> wrote:
Hi, all. I am from Singapore, we deployed a few PABX based on Asterisk.
Here in Singapore there are two Teleco providing E1 pri service, we
encountered a strange problem : when calling a number that is
unavailible or line suspended, one of the
I've solved the problem, but still not understanding very well why do i need it:I've inserted inside [ext-did-custom]exten=>h,1,hangupWhy do i need this? this is not usually used to run something after an hangupcall?
thks!On 9/5/06, Marco Mouta <[EMAIL PROTECTED]> wrote:
Hi all,I think i'm missing
If it shows in the show modules command, it means, the module is loaded,
right?
If yes,
^CLI>show modules like app_re
Module Description
Use Count
app_realtime.soRealtime Data Lookup/Rewrite 0
app_readfile.so
"Llorenç Suau" <[EMAIL PROTECTED]> writes:
> Any suggestions, to how I can make that the PBX receives correctly the call,
> PREFIX+number, to make the external call.
Does this link have the right to make calls to the outside world on
the PBX? Normally this feature is turned off on typical PBX.
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