On 19/12/06, Doug Crompton [EMAIL PROTECTED] wrote:
Is what I am trying to do in this context possible. That is changing the
incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not
preceeded by a 1 I want to add a 1. Often calls come in without the
preceeding 1 and this plays havoc
Hi Paul Eric,
Thank you for you information and quick response. I had enabled Monitoring in
every SIP phone already. Made some Playback see below truncated config;
exten = s,21,Dial(${pre_num:4}${the_num}${post_num},120,${TRUNK_OPTIONS})
exten = s,22,Goto(s-${DIALSTATUS},1)
exten =
Hi all,
I am having a couple of problems with features.conf I was hoping to get
some help with.
#1. If an outside caller is parked, when retrieved, that caller will
now have the ability to transfer. This only happens when they are put
in call parking and then retrieved.
#2. I cannot
Hey all,
I am very interested in using AEL2 (don't want to upgrade to 1.4 to get
it though), but am having some problems upgrading/patching my asterisk
system. I am following the instructions on the wiki:
http://www.voip-info.org/wiki/view/Asterisk+AEL2#AEL2AnnouncementsandNews
But get
Am Dienstag, den 19.12.2006, 01:13 -0800 schrieb Angel Heart:
Hi Paul Eric,
Thank you for you information and quick response. I had enabled
Monitoring in every SIP phone already. Made some Playback see below
truncated config;
exten = s,21,Dial(${pre_num:4}${the_num}${post_num},120,
Am Dienstag, den 19.12.2006, 01:11 -0500 schrieb Doug Crompton:
Is what I am trying to do in this context possible. That is changing the
incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not
preceeded by a 1 I want to add a 1. Often calls come in without the
preceeding 1 and
Hi all,
What will happen if 2 devices using the same set of sip account to
connect to the same asterisk? Do they both can make call? Can they
receive call as normal?
___
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asterisk-users mailing
Hi
On 12/19/06, Danny [EMAIL PROTECTED] wrote:
Hi Hermann !
I am using this script [ check the commented line ]
Can we please stay within the topic of this thread?
Thanks
JY
___
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Hi,
I want to unsubscribe from asterisk-users-request-lists, and donot
want to recieve mail any more.
Kindly unsubscribe me...
sanchal singh
On Mon, 2006-12-18 at 13:57, [EMAIL PROTECTED]
wrote:
Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com
To
Hi,
It seems that they both can make calls, but only one can receive call: the
last registered...
Greg
Hi all,
What will happen if 2 devices using the same set of sip account to
connect to the same asterisk? Do they both can make call? Can they
receive call as normal?
Hi Lex,
Ok, so I switched the Sangoma for a Digium Quad E1 card, but still now luck.
Here is my config, can you spot my mistake:
zaptel:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone=uk
defaultzone=uk
zapata:
immediate=no
switchtype=euroisdn
signalling=pri_cpe
group=1
Hi ,
Has anyone earlier tried integrating asterisk with LDAP.
I am interested to integrate LDAP for authentication purpose for any SIP
Incoming calls..
Pl. suggest pointers.
Thanks and Regards
--Sandeep Kalra
Ph: +91-120-4342000-X-2966
:
I have a line from BT (UK) connected to my asterisk system, on a
TDM400P.
I am able to see either distinctive ring cadences or caller ID but not
both. If I try to enable both, all drings show up as 0,0,0.
This is a pain because, if I make a call out over that line and the
number I call is busy,
On Tue, Dec 19, 2006 at 11:22:53AM +, Phil Reynolds wrote:
I have a line from BT (UK) connected to my asterisk system, on a
TDM400P.
I am able to see either distinctive ring cadences or caller ID but not
both. If I try to enable both, all drings show up as 0,0,0.
This is a pain
19 dec 2006 kl. 11.58 skrev Gregory Duchatelet:
Hi,
It seems that they both can make calls, but only one can receive
call: the
last registered...
Greg
Hi all,
What will happen if 2 devices using the same set of sip account to
connect to the same asterisk? Do they both can make call?
On Tue, Dec 19, 2006 at 01:45:00PM +0200, Tzafrir Cohen wrote:
Don't know about distinctive ring. As for caller ID:
Have you set zapata.conf to use v23 signalling for callerid?
callerid=asreceived
cidsignalling=v23
cidstart=polarity
Yes - and it works, but breaks distinctive ring
2006/12/19, C F [EMAIL PROTECTED]:
Can anyone recommend a call accounting solution with rating for post
paid billing that works well with asterisk using the account code or
any other info from the CDR?
I don't want the billing software to any phone calls for me, therefore
any solution that
So I moved the Ethernet controller to IRQ 11 and the 'Unknown Device' followed!!
I'm kind of at a loss with this machine, as I don't normally deal with
IBMs. Here is the full output from the command.. can someone point
out where the Digium card is, because I don't see it.
[EMAIL PROTECTED] ~]#
hello,
i try to build a IVR for our company my problem is that the dtmf tones
are not recognized by the phones i tried several phones.
BUT when i call the voicemail i can navigate with all phones through the
menu. I use * 1.2
here is the context:
[ivr]
exten = s,1,Answer
exten =
One thing I've noticed, is any time I make changes to Asterisk I have to
reboot the phones to keep BLF up to date. Have you tried that?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Johnson
Sent: Monday, December 18, 2006 6:07 PM
To:
Or web into the phone and click any submit button - not a great idea
though if you remotely provision, just make sure you do not change
any settings as they will then over ride the remote file settings
On Dec 19, 2006, at 1:09 AM, Douglas Garstang wrote:
From the Asterisk console:
sip
Could the unknown device be a management card?
The newer dells have a management card built into the fist ethernet controller.
--
--
Steven
http://www.glimasoutheast.org
Matt [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
So I moved the Ethernet controller to IRQ 11 and the
Google: asterisk ldap
http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+ldap.conf
Never done it myself though.
Steve
On 12/19/06, sandeep kalra [EMAIL PROTECTED] wrote:
Hi ,
Has anyone earlier tried integrating asterisk
Hi,
I need to connect to a remote VOIP server that only uses G.729 codec.
From our Asterisk server, we will then make several calls ( 1 but
?? !!) in the same time to the remote VOIP server.
Do we need to purchase Asterisk G.279 license ? If yes, how many
licenses must we buy?
Thanks
OK with the remote server on one side doing G729, what will you be
connecting to on the other side? If it does G729 then no license, if
not then one license per active call. Also if * will be doing any
voicemail etc then you will also need the license.
On Dec 19, 2006, at 8:31 AM, Michel
(1st attempt was rejected by postfix @lists.digium.com, here goes the 2nd)
-- Forwarded message --
From: Ex Vitorino [EMAIL PROTECTED]
Date: Dec 18, 2006 11:41 PM
Subject: Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME
To: Asterisk Users Mailing List -
Hi can i do that on asterisk?
- receive a call from h323
- call an internal sip extension AAA
AAA does meetme on
SIP/[EMAIL PROTECTED]
SIP/[EMAIL PROTECTED]
Finally, the caller from h323, user1 and user2 can speak together...
Is it possible?
Many thanks..
So I moved the Ethernet controller to IRQ 11 and the 'Unknown Device' followed!!
I would try moving the Digium card to another slot. Your Ethernet
controlled must be onboard and it share its IRQ with the slot where
the Digium board is.
hth
___
Hi,
I was trying out call file just to see how they worked and my system
does not seem to do anything with them, although asterisk *is* deleting
the files that I put into /var/spool/asterisk/outgoing.
1. I nano'd a quick call file like so:
Channel: SIP/axVoice/910555
CallerID : Leebo
I spent hours debugging this a few weeks ago.
The ${UNIQUEID} contains a period (.). Mine are something like
.xx. When soxmix is executed to mix the in and out files, the
file types are not specified. This causes soxmix to attempt to
determine the file type by the filename's
Hey all... Scenario
(INTERNAL)
1 Call comes in to receptionist and gets transferred to someone
2 No one picks up that transfer
3 Call goes back to receptionist
Now when the call goes back to the receptionist, how can I change either
the ringer, the callerID or both?
* TIA
On Tue, Dec 19, 2006 at 10:17:51AM -0500, Lee wrote:
Hi,
I was trying out call file just to see how they worked and my system
does not seem to do anything with them, although asterisk *is* deleting
the files that I put into /var/spool/asterisk/outgoing.
1. I nano'd a quick call file
How would I parse the area code from this variable? Number=2515551212
Sorry for the dense question, I don't seem to be able to find an
appropriate function for parsing left to right.
___
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Ok so that 'unknown' is infact the Digium card then? I suspected that.
On 12/19/06, Time Bandit [EMAIL PROTECTED] wrote:
So I moved the Ethernet controller to IRQ 11 and the 'Unknown Device'
followed!!
I would try moving the Digium card to another slot. Your Ethernet
controlled must be
Ok does anyone know of any softphones that will dial DTMF tone keys A B
C D
And do you know if Asterisk will take the DTMF Tones for A B C D
--
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VoIP PBX) 1-563-773-6610 EXT: 250
-- For Information on PBX
Does anyone know how to remotely reboot a PolyCom specifically 601
phone?
sip notify polycom-check-cfg ipaddr
Or you might have to switch the polycom-check-cfg and the ip. I forget the
order. You
also need to make sure that the phone has alwaysreboot=1 in the sip.cfg xml
file.
You can
On Tue, 19 Dec 2006 10:35:32 -0500
Matt [EMAIL PROTECTED] wrote:
Ok so that 'unknown' is infact the Digium card then? I suspected
that.
The Vendor ID is 'D161' which is supposed to look a bit like the first
four letters of 'Digium' :)
gdh
___
On 12/15/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
nik600 wrote:
Hi
i am trying to do the same thing:
receive a call from a cisco callmanager and forward it to a SIP user.
Asterisk is compiled with h323 support, and is configured as a gateway
in the cisco callmanager.
The incoming call
Idefisk will do that - www.asteriskguru.com . (And asterisk will accept it).
Zoa
Al Bochter wrote:
Ok does anyone know of any softphones that will dial DTMF tone keys A
B C D
And do you know if Asterisk will take the DTMF Tones for A B C D
___
${Number:-10:3} if I recall correctly would give you 3 characters
starting at the 10th from the end.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of John French
Sent: Tuesday, December 19, 2006 10:35 AM
To:
Dear asterisk users,
I am using Asterisk and I a m a new user.
Before it was working properly.
Since two days, users can not get registered : users registered timeout.
Those are the results of commands
1. /var/log/asterisk#asterisk-rvv
== Parsing
It seems that Greg is truth for the case. Asterisk doesn't care how
many devices register to the same account as it is a feature of sip
protocol (please let me know if there is a method to restrict it).
In my case, I use a soft phone an hard phone using the same sip
account information to
Now for some reason instead of giving me an error on the caller ID, it's
not mentioning the caller ID at all. Is there some explicit thing I
need to put in to get the caller ID?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
On Tue, Dec 19, 2006 at 09:32:00AM -0500, Steven wrote:
Could the unknown device be a management card?
The newer dells have a management card built into the fist ethernet
controller.
D161:2400 is a Digium TDM2400P card.
Source: Debian has a nice 'update-pciids' command, which updates the
On Tue, 2006-12-19 at 04:20:07 -0500, [EMAIL PROTECTED] wrote:
Hey all,
I am very interested in using AEL2 (don't want to upgrade to
1.4 to get
it though), but am having some problems upgrading/patching my
asterisk
system.
It doesn't have anything to do with hardphone versus softphone. The
issue is that it can only keep track of one registration per account.
When the hardphone gets unplugged, it will not know about the softphone
until it registers with asterisk. It's initial registration was lost
when the
John French wrote:
How would I parse the area code from this variable? Number=2515551212
Sorry for the dense question, I don't seem to be able to find an
appropriate function for parsing left to right.
___
--Bandwidth and Colocation provided by
Do you, Gordon or Doug, happen to place international calls with
early-dial enabled? What kind of extensions.conf magic do you work to
allow this?
I have been trying for some time to get this to work. (My message from
2006.11.03 regarding this is quoted just below)
On 11/3/06, *Anthony
I'm wondering if moh is still broken in Asterisk 1.4 beta3. In Asterisk 1.2,
when a callee put a caller on hold, the musiconhold class that was played was
not the one the callee wanted the caller to hear, but something else. Even
after using mohsuggest in Asterisk 1.4, it still appears that
Hi -
(INTERNAL)
1 Call comes in to receptionist and gets transferred to someone
2 No one picks up that transfer
3 Call goes back to receptionist
Now when the call goes back to the receptionist, how can I change either
the ringer, the callerID or both?
We'll need to see a little more info to
In the CLI:
sip show peer axVoice
show dialplan main_menu
set verbose 3
Then drop the call file
What is the CLI trace of the above?
Hi, thanks for responding. Please see the output below.
Please note that moving a call file into /var/spool/asterisk/outgoing
did not produce any CLI
Your phones only register once, when they first start up. Seems to me that
having multiple phones on the same account is asking for trouble- why not set
up multiple accounts in the usual way, and create a ring group for all the
phones you want to use? Like this example that rings two phones at
how isit possible to get the VM there when one line is busy?
regards rene
On Tue, 19 Dec 2006 09:48:01 -0800
Carla Schroder [EMAIL PROTECTED] wrote:
Your phones only register once, when they first start up. Seems to me that
having multiple phones on the same account is asking for trouble- why
Early dial is a real nice feature BUT it requires that you carefully plan
and design your extensions. Each digit is accepeted by Asterisk and if a
match exists up to that point it will be accepted and dialed.
As an example, my internal extensions are 4xx and my internal special
extensions are
Thanks Anselm, That did it!
Doug
On Tue, 19 Dec 2006, Anselm Martin Hoffmeister wrote:
Am Dienstag, den 19.12.2006, 01:11 -0500 schrieb Doug Crompton:
Is what I am trying to do in this context possible. That is changing the
incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is
Noah Miller wrote:
Hi -
(INTERNAL)
1 Call comes in to receptionist and gets transferred to someone
2 No one picks up that transfer
3 Call goes back to receptionist
Now when the call goes back to the receptionist, how can I change either
the ringer, the callerID or both?
If you're looking
I understand how early dial works (484 response and all that jazz), I
also understand the NANP and how to keep my extensions from
overlapping... but thank you for the tips.
My question was: Do you place international calls from phones with
early-dial enabled?
If so, might you be willing to
On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote:
In the CLI:
sip show peer axVoice
show dialplan main_menu
set verbose 3
Then drop the call file
What is the CLI trace of the above?
Hi, thanks for responding. Please see the output below.
Please note that moving a call
Ok.. so then for some reason the PCI slot that the Digium card is in
is following the IRQ of the Ethernet controller. We will move the
Digium card and see what happens.
On 12/19/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Dec 19, 2006 at 09:32:00AM -0500, Steven wrote:
Could the
Hi Rene -
how isit possible to get the VM there when one line is busy?
If I understand your question correctly, the answer is you need two
incoming phone lines.
- Noah
___
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asterisk-users
Sorry, I did not read the original message completely. The answer is no I
do not make international calls. I do not know anyone in any other country
to call! I do not have a rule for that but it should be easy to implement
as 01x would not match anything I currently have for early dial. Would you
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bruce Ferrell
Sent: Tuesday, December 19, 2006 12:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Parsing Area Code from CallerID
Hmm, I don't know what happens when one of the lines is busy and none of the
lines get answered. It's easy enough to test. If it doesn't go to voicemail,
then perhaps this is what you want:
http://www.voip-info.org/wiki/view/Asterisk+tips+findme
On Tuesday 19 December 2006 9:58 am, René Enskat
Tzafrir Cohen wrote:
On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote:
In the CLI:
sip show peer axVoice
show dialplan main_menu
set verbose 3
Then drop the call file
What is the CLI trace of the above?
Hi, thanks for responding. Please see the output below.
Please note that moving a
Are you kidding? Lighten up people!
Al made a friendly recommendation based on the comments regarding TrixBox.
Go have a beer... take a load off... enjoy the holidays.
Regards,
David
___
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Steve Murphy wrote:
Lee, everyone--
Sorry about that. I've created a patches subdir in the AEL2-1.2
repository, and put that patch down in there. So, now, you do:
svn co http://svn.digium.com/svn/asterisk/team/murf/AEL2-1.2/patches
and then
Excellent! Thank you. I will try this.
Again,
I am located on the west coast of the united states.
In order to dial an international number from within the US, we must
first dial the special international access code that tells the PSTN
the following call is an international one - in the US that is 011,
followed by the country code, and
nothing happen it only let ring all lines which are not in use but i want that
the busy vm message is coming when one line is busy.
On Tue, 19 Dec 2006 10:55:34 -0800
Carla Schroder [EMAIL PROTECTED] wrote:
Hmm, I don't know what happens when one of the lines is busy and none of the
lines get
Hi -
(INTERNAL)
1 Call comes in to receptionist and gets transferred to someone
2 No one picks up that transfer
3 Call goes back to receptionist
Now when the call goes back to the receptionist, how can I change either
the ringer, the callerID or both?
If you're looking for the
We use this function regularly (you should see my phone's
dialstring...). If one phone responds that it's unavailable, the rest
of the phones will still ring through. In the event that none of the
other phones are answered, the extension is considered unanswered, so
depending on how you program
Lee wrote:
As I mentioned above, the action of dropping a .call into the
/outgoing directory did not produce any CLI output. I did this
through 2 putty sessions. The first, we setup to watch the CLI output
and the second was to use the commandline to move the .call into the
/outgoing
yes thats we i use 100,a,b,c etc.
do you can mail me your extensions how you do the dials and the vm?
regards rene
On Tue, 19 Dec 2006 13:15:53 -0600
Aaron Daniel [EMAIL PROTECTED] wrote:
We use this function regularly (you should see my phone's
dialstring...). If one phone responds that
Are you sure IdsFisk will do all 16 DTMF tones?
I have the free ver of IdeFisk and that one does only the base 12 DTMF tones
Base 12 DTMF are
1 2 3 4 5 6 7 8 9 0 # *
The 16 DTMF are
1 2 3 4 5 6 7 8 9 0 # * A B C D
If the paid ver of IdeFisk has that may have to pay the money but first
I must
If you are using Windows to generate the .call files, make sure they are in
Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files. Use
Crimson Editor www.crimsoneditor.com to make the file, and click Document
File Format Unix Format.
I ran into this same problem, and it turns
Noah Miller wrote:
Hi -
We'll still need to see more of your dialplan. By your description,
it looks like the call is failing because the Dial() times out.
Take two... My calls are NOT FAILING. Never have so let me restate...
Call comes in receptionist answers. For some ungodly reason this
Does the script run from command-line? Without taking a close look at
this, the include statements in the function body of connect_db look
potentially messy.
Also, any output to stdout is interpreted by asterisk as a command, so
those fputs statements would be a problem -- do
The free version 1.31 has all 16 keys on the keypad.
Bob...
Al Bochter wrote:
Are you sure IdsFisk will do all 16 DTMF tones?
I have the free ver of IdeFisk and that one does only the base 12 DTMF
tones
Base 12 DTMF are
1 2 3 4 5 6 7 8 9 0 # *
The 16 DTMF are
1 2 3 4 5 6 7 8 9 0 # * A B C
Hmm, if the latest free version does not have all 16 keys, email
[EMAIL PROTECTED], there should not be a difference in the amount
of DTMF keys between biz and free version.
Zoa
Bob Chiodini wrote:
The free version 1.31 has all 16 keys on the keypad.
Bob...
Al Bochter wrote:
Are you
On 12/17/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Dec 17, 2006 at 07:07:07PM +0200, Maxim Vexler wrote:
Hi list,
It's been a while since I've done asterisk stuff, and I'm wondering if
there any news in the field.
What do you people use today for http management of debian based
Colin Anderson wrote:
If you are using Windows to generate the .call files, make sure they are in
Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files. Use
Crimson Editor www.crimsoneditor.com to make the file, and click Document
File Format Unix Format.
I ran into this
On 23:33, Tue 19 Dec 06, Maxim Veksler wrote:
Are there any other tools then, perhaps some that has not been debianized
yet?
I'd like something that could be more cooperative with user hand made
changes.
Thanks to the folks at SpeakUp (http://www.speakup.nl) I
have a nice webtool that puts
Anyone know if there's a way to match a dialplan extension, execute some code,
say set a variable, and then continue with the dialplan?
I want to set a variable when the dialplan flows beyond a certain context. This
would be a great feature.
Doug.
Hi all,
I've got Asterisk 1.2.10 up and running on Debian using the back ports.
I noticed that it didn't come with mpg123 or depend on it and I believe
I read somewhere that asterisk now handles it's own mp3 playback? Is
this true? If so I must have a problem, because I hear no music when
On 12/19/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Anyone know if there's a way to match a dialplan extension, execute some code,
say set a variable, and then continue with the dialplan?
I want to set a variable when the dialplan flows beyond a certain context. This
would be a great
I downloaded the free 1.37 version. The slide out keypad displays ABCD
buttons but they do not respond to clicks. You can enter ABCD using your
keyboard and they will be sent to Asterisk.
There appears to be some funkyness with case though.
If you enter and click dial, is sent to
When setting up the zaptel, etc Is it necessary to have a seperate
group for incoming and one for outgoing calls? Either way, does asterisk
always know which channels are open, and does it always clear up a
channel for use once a call completes?
Reason for asking... After dialing into our
I just know someone is going to ask 'why would you ever want to do that?'.
Here's my answer.
We have two companies, each with a dialplan similar to what's below. In the
event that the number being dialled does not match any number within our OWN
company, we want to set the caller id to be a
I had this same problem. I also read that mpg123 was not required, but it
actually is if you wish to use mp3 files. I just decided to go with RAW files
because I had problems converting some mp3's to the appropriate bit rate.
-Original Message-
From: David Thomas [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 3:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
On 12/19/06, Douglas Garstang [EMAIL
Short answer: a single group should be fine. Long answer: It depends.
Your Dial() command determines the order in which Asterisk plucks channels
from your PRI. Most north american system call inbound channel 1 first, then
2, etc. It makes sense, then for you to take channels from the topmost
Now for some reason instead of giving me an error on the caller ID, it's
not mentioning the caller ID at all. Is there some explicit thing I
need to put in to get the caller ID?
callerid=asreceived
___
--Bandwidth and Colocation provided by
The only other thing that comes to mind is that .call files are very
sensitive to whitespace; you may have unintentially padded the .call file
with whitespace or tabs that it does not like.
The attached .call file works on my 1.0.9 server. Maybe it can give you some
insight.
-Original
On Tue, Dec 19, 2006 at 04:49:36PM -0500, Phil Finkler wrote:
Hi all,
I've got Asterisk 1.2.10 up and running on Debian using the back ports.
Debian does not include the default MoH files that come with Debian for
legal reasons. Get some sound files in the moh directory, basically, and
OK. My basic asterisk install seems to be working. I can get caller
ID. My dialplan says:
[incoming]
; incoming calls from the FXO port are directed to this context from
zapata.conf
exten = s/9185415897,1,Set(CALLERID(name)=Michael Sullivan)
exten = s/9185415897,1,HANGUP(1)
exten =
On Tue, Dec 19, 2006 at 11:33:55PM +0200, Maxim Veksler wrote:
On 12/17/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Dec 17, 2006 at 07:07:07PM +0200, Maxim Vexler wrote:
Hi list,
It's been a while since I've done asterisk stuff, and I'm wondering if
there any news in the field.
a2billing
Is very good
On 12/19/06, Giedrius Augys [EMAIL PROTECTED] wrote:
2006/12/19, C F [EMAIL PROTECTED]:
Can anyone recommend a call accounting solution with rating for post
paid billing that works well with asterisk using the account code or
any other info from the CDR?
I don't
On Tue, 19 Dec 2006, Carla Schroder wrote:
Your phones only register once, when they first start up. Seems to me that
having multiple phones on the same account is asking for trouble- why not set
up multiple accounts in the usual way, and create a ring group for all the
phones you want to use?
I am running console. I'm a newbie for AGI's but not that new.
Thanks,
bp
On 12/19/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Dec 19, 2006 at 09:35:56AM +0200, yusuf wrote:
to see debug output for AGI's, you *must* be connected to the first Ast
terminal. So start Asterisk like
On Mon, Dec 11, 2006 at 06:48:18PM -0500, Andrew Joakimsen wrote:
You need to understand how NAT works, if you can chan2 and chan2 is behind a
NAT and suddenly someone else is invited to chan2's IP address port 5060
chan2's router willl say WTF I dont have an estabished connection on port
5060
On Tue, 19 Dec 2006, Anthony Kepler wrote:
Do you, Gordon or Doug, happen to place international calls with early-dial
enabled? What kind of extensions.conf magic do you work to allow this?
I have been trying for some time to get this to work. (My message from
2006.11.03 regarding this is
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