I just downloaded the new astcc and it includes now a new field in the
list of the cards: Brand
Great!
How can I use it in the dialplan?
bye
Ronald
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Matthew Boehm wrote:
You may not have most recent CVS. You should have this in your sip.conf:
You are right, ... but the sip.conf will not be updated anyway, if I do
not want to loose all my settings.
rtcachefriends=yes
; Cache realtime friends by adding them to the internal list
; just like
James Taylor wrote:
Just installed ASTCC, got it working.
I've noticed that only part of the sounds come from Allison.
Someone (male voice) has recorded the necessary balance,call cost, etc.
So, there's this mix of male/female announcements.
Is this new or am I missing some sound files?
It seems
Matthew Boehm wrote:
Are you sure that NAT is set correctly everywhere? I sometimes forget to set
the phone to be NAT aware.
That is weird that 'sip show peers/users' doesn't show the phone both times.
Have you stopped/started asterisk since these changes? Do it again just to
make sure.
The only
/configs/ that you can read through.
thanks, I did not notice that!
bye
Ronald
-Matthew
From: Ronald Wiplinger [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Mon, 14 Mar 2005 09:27:52 +0800
To: Asterisk Users Mailing List
I am trying to get more familiar with ASTCC, but I miss some tools, but
I believe somebody has already thought about it:
1. I would like to send a standard letter to the users, as soon their
balance drops below a certain value. E.g., Dear user, you have only 1.-
left, consider to fill
I am checking on the SetGroup / CheckGroup commands, but I have some
troubles to undestand the examples.
SetGroup(moh) can be moh anything as I like? Usually moh stands for
music on hold
CheckGroup(1) checks if somebody in in group moh. Does it mean I can
only have one SetGroup(xxx) ??
Matthew Boehm wrote:
INSERT INTO sip_buddies VALUES
(1,'621',NULL,NULL,NULL,'\Demo\,621','yes','inhouse',NULL,'rfc2833',NULL
,N
ULL,'dynamic',NULL,NULL,NULL,NULL,'[EMAIL PROTECTED]',NULL,'yes',NULL,NULL,NULL,'1',
''
Ronald Wiplinger wrote:
I am trying to figure out a way to add something like:
61 100 pennies (Everything what is not
listed below)
61 78150
61 5 130
61 342 180
How could I do these (four) regex?
Since nobody answered to my
It bothered me, so I changed it to my need, how can I contribute this
changes back to the community?
I copied both *.cgi of ASTCC into new ones, added the start time into
the table
I tried diff oldfile newfile, but it does not look like then normal
patches, ... how should I make it? Where to
I just downloaded the new astcc and it includes now a new field in the
list of the cards: Brand
Great!
How can I use it in the dialplan?
bye
Ronald
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, if it just a word!!!
I would need that a user could choose between two tarriffs, ... I
thought that would be great to use Brands for that.
bye
Ronald
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Nabeel Jafferali wrote:
I would need that a user could choose between two tarriffs,
... I thought that would be great to use Brands for that.
Like I said, a brand is used when you are creating a card. Therefore,
the markup defined by the brand is applied to the card. Simple?
Nabeel
I think
Ronald Wiplinger wrote:
Matthew Boehm wrote:
INSERT INTO sip_buddies VALUES
(1,'621',NULL,NULL,NULL,'\Demo\,621','yes','inhouse',NULL,'rfc2833',NULL
,N
ULL,'dynamic',NULL,NULL,NULL,NULL,'[EMAIL PROTECTED]',NULL,'yes',NULL,NULL,NULL,'1',
''
,'999',NULL,NULL,NULL,'Password','friend
I have setup flash pannel, ... looks nice, but so far I could not
configure it to get more than 4x7 buttons.
I tried to make the buttons smaller, but than just the entire picture is
smaller.
The description says you can have a hundred buttons,
Can I have multiple flash pannels? E.g. for
Matthew Boehm wrote:
Is there anything I can do to track down the problem? e.g., is there a
command in *CLI to read the database record? push a record, see the
differences, ...
You can try this:
realtime update sippeers allow g729 name 621
That should be the SQL equivalent to UPDATE TABLE
Martijn van Oosterhout wrote:
On Wed, Mar 16, 2005 at 03:25:17PM +0800, Ronald Wiplinger wrote:
Mar 16 15:13:45 DEBUG[29502]: Raw Hangup 69.73.19.178:4569, src=14, dst=1259
Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Update SQL: UPDATE
sip_buddies SET name = '621' WHERE allow = 'g729'
Mar 16
Matthew Boehm wrote:
Ronald Wiplinger wrote:
[mysql1]
dsn = astconf
username = root
password = MyPassword
pre-connect = yes
You are not using the ODBC drivers. You can remove that [mysql1] stuff
from your res_mysql.conf
Removed, but still no codecs
br
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I tried wiki, but I got too many pages (I think all of them), ...as answer.
I want to write an agi.
I need a HOW-TO, is there anything available?
bye
Ronald
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What does that mean? Where can I get more info about it?
Mar 18 07:19:51 WARNING[9563]: chan_sip.c:7549 handle_response: Got 200
OK on REGISTER that isn't a register
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I had the impression that the command:
*CLI realtime load sippeers name 621
(The new configuration was displayed after that command)
would re-load the config of phone 621
(I changed the context and tried above command, however, it kept the old
info!)
bye
Ronald
Vyom A wrote:
In X-Lite, the Conference button is disabled, but that can probably
be done in X-Pro (from the XPRO_users_guide.pdf)
There is a crack available:
http://www.regnow.com/softsell/nph-softsell.cgi?item=9054-12
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Matthew Boehm wrote:
Ronald Wiplinger wrote:
I had the impression that the command:
*CLI realtime load sippeers name 621
(The new configuration was displayed after that command)
would re-load the config of phone 621
I may be wrong but realtime load is simply a debug tool for CLI so
After I have finished to key in the area codes for Mexico I would like
to propose:
The guy how created the numbers should be stoned to death with the dice
he created the numbers !!!
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PS: Spam
,
MATT---
P.S. Yes, I also think they could've done this much better, but what do you
expect from bureaucrats.
-Original Message-
From: Ronald Wiplinger [mailto:[EMAIL PROTECTED]
Sent: Friday, March 18, 2005 7:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
I finished my setup for ASTCC and I am looking for a tool to convert a
mysql table to excel and back.
Which one is good, and where can I find it?
bye
Ronald
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How to install /use festival on Asterisk?
I would need text to speech in:
English
German
andChinese (Mandarin)
bye
Ronald
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To
I try to change something in ASTCC, but I am now totally blind,
I hang on one line now. I changed:
vpbx:/var/lib/asterisk/agi-bin # diff astcc-original.agi astcc.agi
22c22
# exten = _00X,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
---
# exten =
I have three different time displays:
Flash panelcaller 615 48:00
called 62058:18
Snom phone shows for the same call 47:55
Why is there a difference at all?
bye
Ronald
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Where can I get a list of all possible SIP ... response numbers and
their meaning?
bye
Ronald
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Words of a user, ... what can I make better?
Most of the calls had a little delay. People on the other end of the
phone said it sounded like cell phone with little stop during the
phone. So, it seems voip is not a good quality pstn phone yet. But I
wondered my classmate that using Dynasky
I am looking for a way to add features to an extension number.
e.g. extension 601 gets features a, b and c, while extension 605 gets
the features a, d and e.
I would like at the beginning query a database to get the flags for the
extension (bellow for 601)
feature_a=y
feature_b=y
feature_c=y
I did not get any hint to my first try, ... can somebody help me?
I try to change something in ASTCC, but I am now totally blind,
I hang on one line now. I changed:
vpbx:/var/lib/asterisk/agi-bin # diff astcc-original.agi astcc.agi
22c22
# exten =
I know that I can use g or G for Zap lines, but how can I use group and
more exactly random lines of a group?
bye
Ronald
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To
How to change the date from:
Mon Mar 23 13:50:43 2005
to:
03-23-2005 09:50:43 Wed
???
1. The form as ASTCC stores the date / time does not allow to sort the
records (ASC/DESC) by date.
I would like to change it to a form that allows me to sort the records.
2. Is there a way to change existing
!
(sorry I could not resist!)
--
Ronald Wiplinger (CEO of ELMIT)
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PS: Spam prevention!
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I have some numbers, which should be treated equally. To avoid double
coding, I would like that this extension could be re-written.
E.g., some users are used to dial 002 ~ 009 as international prefix,
while I have choosen to use the USA way (011).
It would be nice if the user can dial
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It just bothers me ;-)
If somebody calls me via PSTN / TDM422 card and hang up, than my phones
still rings!
What do I need to change for that?
exten = s,1,NoOp,${CALLERIDNUM}-${CALLERIDNAME}
exten = s,2,Answer
exten = s,3,zapateller ; torture telemarketers
exten =
I found is not because of time drift, ... it seems to
be constant 10 minutes !
4. Can I have overlapping multiple pages?
E.g. a customer of mine will see only HIS extension numbers, but I can
see all?
bye
Ronald
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http://www.elmit.com+886 (0) 939--77-55-16
I want to change the below lines:
exten = _011.,1,SetGroup(line1); set current group to line
exten = _011.,2,CheckGroup(1); check line1 does not
have more than 1
exten = _011.,3,Dial,SIP/[EMAIL PROTECTED]; use line-1
exten =
Matthew,
I remember somewhere I read that you have setup additional fields in
your database (sip-buddies).
Do you have an example how to use it?
I am looking for a way to add / allow features to certain users.
bye
Ronald
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I cannot figure out why some phones need about 11 seconds till they ring.
Both phones are on my desk. I call from A to B. Other phones ring instantly!
What can be the reason for that?
bye
Ronald
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Andres wrote:
Tim Connolly wrote:
Can anyone suggest a way to randomly pick an outbound trunk so that
all trunks are used equally? Im all about Equal Trunk Opportunity
Seriously though, Ive got 500 minutes per softphone account and Id
like to use an equal amount on each as to not see overages
Not really an Asterisk questions, but I noted a lot of subject lines,
which are NOT very attractive to open the message.
These are messages, like:
Can * do that?
Is this possible?
Can anybody help me?
I got a problem (again)!
We are all busy and we all like to know what to do better and share
You can't see the sweat, but ...
I would like tp post my improvements to ASTCC somewhere, ... but where???
bye
Ronald
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I put the Who? in Mishehu wrote:
I'd put the device and another machine on a separate physical network
where you can make whatever IP configurations you need in order to be
able to send data to the IAXy. Then you can load new configuration to
it there.
There might be a better way to do i, but
I would like to get a notice by email, if we run out of gateways!
exten = _9011Z.,410,Busy
exten = _9011Z.,411,EMAIL = How to?
bye
Ronald
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I tried to use ONE entry of my voicemail.conf to put into the database:
[other]
;602=1357,Ronald Wiplinger 602,[EMAIL PROTECTED]
INSERT INTO `voicemail_users` ( `uniqueid` , `customer_id` , `context` ,
`mailbox` , `password` , `fullname` , `email` , `pager` , `stamp` ,
`attach` , `saycid
Darren Wiebe wrote:
That capability is not there yet. I would personally recommend
using the 'Local' channel and routing your calls via the
extensions.conf file. This is totally up to you but I find it gives
me more flexibility. That would also make it easier to do something
like you are
Matt Schulte wrote:
Now, this has been answered many, many, many times...in fact..I
believe Olle answered this in his Welcome to Asterisk post he sent out
over the weekend.
Hmm, into my mailbox it did not come, ...
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Laurent Foulonneau wrote:
Hello list,
Does anyone know about a web/php interface to deal with users in
Realtime's Mysql database (sipusers and sippeers tables) ?
Thanks in advance
Laurent
PhpMyAdmin
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We use wakeup calls for reminders, but it happens, that the person to be
reminded is on the phone. To get a voicemail later is not really useful
anymore, ...
Is there a way to avoid that?
bye
Ronald
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Steve Edwards wrote:
On Thu, 7 Apr 2005, Jason Brown wrote:
Does anyone have a working failover outbound calls that I could sponge a
hint from? i.e.
Exten = _1NXXNXX,1,Dial(IAX/MyFirstCrappyIAXPeer/${EXTEN},60)
Exten =
Matt Riddell wrote:
Matt Schulte wrote:
Is there an SRV bounty out there yet? $500 to the first person who
implements it (correctly :-) )..
Once somebody told me, if you do not know what it is, you most likely do
not need it.
However, I can hardly follow that advice. What is SRV?
bye
Ronald
Guillermo Salas M wrote:
On Fri, 2005-04-08 at 02:57, San Singhania wrote:
Hello Asterisk community,
After numerous request from various companies where we have
implemented * as a phone system and also
from many other * users all over the world, yesterday we released the
1st version of
I should connect to a gateway and got following info:
Username = Password = NONE(not very secure!!!)
SIP
port 5060
IP address
For a trunk line dial 1234 and continue the number you want to reach at
PSTN.
codex g723 (I guess it should be g723.1)
vpbx*CLI
-- Executing NoOp(SIP/615-127a,
What does it mean, and how can I fix it?
Apr 8 23:50:23 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown
SIP command 'PUBLISH' from '192.168.250.108'
Apr 8 23:50:24 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown
SIP command 'PUBLISH' from '192.168.250.108'
Apr 8 23:50:24
Chris Mason (Lists) wrote:
Call Accounting is such an important issue for me it is literally a make or
break component, without it I will not be able to deploy Asterisk at our
resort. If I have to use a windows computer to download and run the client
end of the software, so be it. At least the
Chris Mason (Lists) wrote:
There are other accountings available, and a Windows solution is for ME
the very last choice!!!
Really? I am about to commission someone to write a call accounting package
for us because there is nothing out there. Exactly what else is available?
I am not sure
Any idea?
-- SIP Seeding peers from Astdb: '3366' at [EMAIL PROTECTED]:64440
for 3600
-- Saved useragent Sipcom/ATA2000-1.6.11 for peer 3366
-- SIP Seeding peers from Astdb: '886229421761' at [EMAIL PROTECTED]:5060
for 3600
-- Saved useragent Grandstream BT100 1.0.5.18 for
Rich Adamson wrote:
Serves you right for offering a bait and switch deal. If you are selling
unlimited that's what it should be. Why would you be surprised if someone
wants to use the unlimited feature?
What's wrong with selling a 1000 minutes for $10 plan? I guess you are
afraid someone will then
Kerry Garrison wrote:
I am trying to put together a matrix. Please send me links, corrections,
additions, flames, etc.
http://www.geekgazette.com/index.php?option=com_contenttask=viewid=25Item
id=26
-Kerry
Kerry,
you did a great job, ... (I made a bookmark of it!!!)
However, I wanted to find
*CLI show version
Asterisk CVS-HEAD-03/13/05-23:38:12 built by [EMAIL PROTECTED] on a x86_64 running
Linux
How can I upgrade safe?
How can I downgrade if something did not work out right?
What should I upgrade?
Where can I read for each package the changes to see if it is worth to
upgrade?
Is
I am lucky that everytime I want to lookup something on the wiki, it is
not available, ... Last time I was lucky to read about faxes, but was
more or less confused about many different approaches to solve it.
What can I get ? (Which feature, which comfort, fax in, fax out, )
I have
I have two snom phones, one is a Snom 190 on my desk and one is a soft
phone snom 360
Extension 615 works fine, but all connections with 616 works not.
On 616 you hear the other party, but the other party get only a white
noise sound !! This is indipendent which direction I call.
What might be
Asterisk is on my box now running about one month without any troubles.
Since two days I got troubles:
1. The Zapta card (2 FXS, 2 FXO) suddenly does not like one phone. It
simple does not supply with a dial tone. You cannot dial. You can reach
it, better say, you can dial it, it rings, but no
for this yet. If anybody is interested in the
project, setting one up for me would be nice :)
Best Regards,
Jean-Michel Hiver.
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Start requires a bit of integration and also requires a few paid
jobs before it can be released as GPL. But it depends on Asterisk::LCR
so I release Asterisk::LCR first. First things first :)
Good job!!!
Thanks - I'll fix the Petal dependency and I'll release 0.03 at some
point :)
--
Ronald
So far I never had problmes with Broadvoice, ...
-- Called [EMAIL PROTECTED]
-- Got SIP response 400 Bad request back from 147.135.4.128
-- Got SIP response 400 Bad request back from 147.135.4.128
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I have some troubles to use Setgroup / Checkgroup!!!
I setup a test (NoOP's are deleted): First caller should get first line,
second caller should get second line, third caller should get busy and
send an email. Note, that I used twice here to check the first line!!!
[trunkint_A]
exten =
I cannot setup the Monitor Extension SIP/650
I tried to use
[650]
username=admin
type=friend
context=inhouse
secret=admin-secret
host=dynamic
nat=yes
canreinvite=no
qualify=yes
[EMAIL PROTECTED]
callerid=ELMIT office,650
callgroup=1
pickupgroup=1
It does not show up as peer.
There is no field in
Good luck!
Ronald Wiplinger wrote:
I have some troubles to use Setgroup / Checkgroup!!!
I setup a test (NoOP's are deleted): First caller should get first
line, second caller should get second line, third caller should get
busy and send an email. Note, that I used twice here to check the
first
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PS: Spam prevention!
Our system is protected with a spam prevention program
Joseph Gutowski wrote:
Are you trying to setup a seperate extension just for IP Switchboard?
That's what it sounds like you're trying to do.
You don't have to do anything to your Asterisk to use the program,
except enable the manager interface and add the 77 and 88 stuff to
your extensions.conf to
Robert Webb wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
dean collins
Sent: Monday, April 11, 2005 5:35 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RE: Ebay listing
Is there a possible settings for a remote SIP phone, so that a router
will not close the connection due to long time inactivity?
bye
Ronald
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Ronald Wiplinger wrote:
Adam Goryachev wrote:
On Tue, 2005-04-12 at 13:40 +0200, Thorben Jensen wrote:
Version 0.80 - 12. April 2005.
* Swedish language added - thanks Daniel Nylander
* Bug fixes
Any chance of integrating some sort of input text box, where you can
just type in the extension
that I can do that from the server
end.
bye
Ronald
On Apr 12, 2005 3:41 AM, Ronald Wiplinger [EMAIL PROTECTED] wrote:
Is there a possible settings for a remote SIP phone, so that a router
will not close the connection due to long time inactivity
I am not sure how many licenses of G729 I need to purchase from Digium.
I have a TDM22 card.
Do I need for each FXS (2) or each FXO (2) or for both the license?
Other SIP phones do have the license already, am I right here?
Thanks for enlighten me here.
Same questions for G723.1 !!!
bye
Ronald
NVC List Manager wrote:
On Friday 08 April 2005 11:57, Ronald Wiplinger wrote:
What does it mean, and how can I fix it?
Use a browser and turn off the Publish request on the Advanced page.
(Obviously you turn the browser to the IP of the phone. See Snom manual for
more help.)
I looked
There are so many fax information available, so that I am getting confused.
What I hope I can get to work:
Any extension should be able to receive fax, whereby via faxdetect the
fax should be sent to the email address as mentioned in voicemail.conf
Which packages should I install?
How would be
, and there should be the answer anyway.
bye
Ronald
On Thursday 14 April 2005 06:14 pm, C F wrote:
Why do ppl do this?
and no I will *not* follow the link.
-- Forwarded message --
From: Ronald Wiplinger [EMAIL PROTECTED]
Date: Apr 14, 2005 7:05 PM
Subject: Please confirm your message
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From: Ronald Wiplinger [EMAIL PROTECTED]
Date: Apr 14, 2005 7:05 PM
Subject: Please confirm your message
To: [EMAIL PROTECTED]
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Daniel Bruce Lynes wrote:
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On Thursday 14 April 2005 19:25, Ronald Wiplinger wrote:
you miss the point.
I have setup the white listing system that I get rid of the spam, and it
works very good.
The message says it very clear what to do, if you cannot
I have put into each phone settings (sip.conf and zapata.conf) in my office:
callgroup=1
pickupgroup=1
I cannot pickup any calls from another phone!!
What do I miss here?
bye
Ronald
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Eric Wieling wrote:
I have put into each phone settings (sip.conf and zapata.conf) in my
office:
callgroup=1
pickupgroup=1
I cannot pickup any calls from another phone!!
What do I miss here?
Your SIP phone is eating the *8. You need to look at your SIP phone
docs, not Asterisk
What am I
I tried many different possible ways to us speed dialing, however, I
end up in the default context, where the number does not match anything,
... with the result Playing 'demo-congrats'
I also could not figure out how to use the tabs Queues and Agents
I have not found a new version over
Ben Price wrote:
Question 1:
If I am going to be selling hardware phones to the enduser do there
accounts have to be SIP or can they still be IAX (I find IAX is better
for firewalls)?
Think about the scale you want to go. If you have many phones you may
want to install a SIP Proxy (e.g. SER)
To complete my project, I would like to setup DIDs in several areas.
What do I need to do that? Another Asterisk box or can I use gateways
instead? Which hardware can I use? Who has experience?
bye
Ronald
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Greg Hill wrote:
On Sat, 25 Dec 2004, Ronald Wiplinger wrote:
To complete my project, I would like to setup DIDs in several areas.
What do I need to do that? Another Asterisk box or can I use gateways
instead? Which hardware can I use? Who has experience?
You either set up your own points
I want to record new sound files in different languages, but I need the
text files of the English ones, which I would use as basic.
Since some languages already exists, I believe such a list should be
exist, but where?
I am planning to make Chinese and Tagalog sound files. If therese are
I have installed Firefly, but I cannot figure out how to use it with
Asterisk.
I have seen the settings in Asterisk, but I do not see any settings in
Firefly.
I need a light
bye
Ronald
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