Do u mean meetme? It is total different from my case.
In meetme, everybody need to know and dial the conference room number
to get into the conference room. In my case, party A,B,C may not know
the conference number. A only knows B numbers and B only knows C
numbers.
On 9/28/07, Pamela Weis
Strange !
We successfully used SuperMicro boards without any IRQ problems.
What is SuperMicro's reply, concerning this IRQ problems ?
They sure have interest to solve this or at least explain why it can't be
done.
Regards
___
Sign up now for AstriCon
Hi,
i think that is not the point.
the call works, what is not working is the IAX config.
somehow i need to put manually all users of the foreign asterisk
(user, password...).
if i put type=friend, it does not work in any case.
if i put type=peer it works only if i define the users of the foreign
Hey folks,
Here's your chance to report in about Astricon, ask or answer general
asterisk questions, talk about your asterisk-related (or voip-related)
projects, sites, work, anything. We interested and listening. We have
a great core group on these conferences, even though Indiana is
This must have been asked before, but googling didn't help much.
How do I define a callerid that contains non-USASCII characters? E.g. ä,
ö, ü, å, ø, æ etc. ?
/Per Jessen, Zürich
--
http://www.spamchek.com/ - your spam is our business.
___
Sign
Thanks.
Actually, I want to have some information about the call transfer just
like to queue_log in queue.
According to your message, there is no such mechanism to associate the
call in call transfer. How about any variable that I can identify the
call which is made by call transfer?
As I know
At 15:37 9/27/2007, Jerry Jones wrote:
I will miss them. It was nice having a local company with a few
Polycoms in stock most of the time. A month or so ago we needed some
quick and were unable to contact them, either through their toll free
or local numbers. I swung by their office last week and
bilal ghayyad wrote:
In the outbound, I read in the documents the Wildcard
match by using the . (period), but I did not
understand how Wildcard will work (like what)?
http://en.wikipedia.org/wiki/Wildcard_character
As I
know that Wildcard is a term used with the Diguim TDM
card (FXO and
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] wrote:
Hello,
I was wondering if it is possible within Asterisk to be in many meetme
conferences at the
same time. This would be sort of broadcasting over all the conferences at
once.
Yes, it is possible, but is a bit fiddly to set up.
A
bilal ghayyad wrote:
Hi list;
While I am writing my configuration on the .conf
files, I would like to know if I wrote the command in
write syntax (form), there is not any way to check if
I am writing correct or not (other than checking my
documentation)?
Also, is there any method for
bilal ghayyad wrote:
Hi List;
In the outbound, I read in the documents the Wildcard
match by using the . (period), but I did not
understand how Wildcard will work (like what)? As I
know that Wildcard is a term used with the Diguim TDM
card (FXO and FXS), so what is the relation between
Douglas Garstang wrote:
Also be sure that you have a very redundant network configuration.
Too often I see people spend a great deal of time and money to get
redundant servers when their switches, firewalls, routers, etc are not
even capable of handling a failed network element.
You can
On Friday 28 September 2007 09:16:14 Rilawich Ango wrote:
Do u mean meetme? It is total different from my case.
In meetme, everybody need to know and dial the conference room number
to get into the conference room. In my case, party A,B,C may not know
the conference number. A only knows B
On Fri, Sep 28, 2007 at 09:57:52AM +0100, Russell Brown wrote:
I've a big problem with SIP forwarding back into 'ringing groups'
creating what can only be described as call storms :-(
I have a 'ringing groups' of SIP phones with an effective dialplan (much
simplified) like so:
;
Hello,
I was wondering if it is possible within Asterisk to be in many meetme
conferences at the same time. This would be sort of broadcasting over all the
conferences at once.
Thanks,
Denis
___
Sign up now for AstriCon 2007! September 25-28th.
Hi List;
In the outbound, I read in the documents the Wildcard
match by using the . (period), but I did not
understand how Wildcard will work (like what)? As I
know that Wildcard is a term used with the Diguim TDM
card (FXO and FXS), so what is the relation between
such cards and the matching in
Hi list;
While I am writing my configuration on the .conf
files, I would like to know if I wrote the command in
write syntax (form), there is not any way to check if
I am writing correct or not (other than checking my
documentation)?
Also, is there any method for searching on specific
topic
On Fri, Sep 28, 2007 at 05:28:21PM +0100, Ade Vickers wrote:
Hi folks,
I've been playing around with an Asterisk server in my office for a few
weeks now, and I've got it pretty much nailed down the way I want it, which
is nice.
One of the features I'm using is the ability to switch
On 9/28/07, William Stillwell (Ki4swy) [EMAIL PROTECTED] wrote:
What is the recommend Digium Card for a PRI in NA ?
William - this has been discussed ad nauseam on the list recently.
Some will suggest that you forget Digium and use instead a Sangoma
card. I personally have only used Sangoma
Hi folks,
I've been playing around with an Asterisk server in my office for a few
weeks now, and I've got it pretty much nailed down the way I want it, which
is nice.
One of the features I'm using is the ability to switch different contexts in
out of the dialplan on a schedule. So, for example,
Another option to you might just be easier. Does your PBX ring your
desk phone for a while and then move on to IVR/auto-attendant? If it
already does, do you have a DoNotDisturb button on your phone? That's
pretty straightforward.
The way we do the switch thing is as follows:
exten =
Hello,
No, in this case wildcard means a symbol that stands for one or more
unspecified characters, used especially in searching text and in
selecting multiple files or directories. There is no relation with the
card which is just a name.
PLL.
Original Message
Subject:
Hi, on 7941G is needful the Call Manager license, the firmware for SIP use
is available (with login) on 7912 and 7940.
Thanks.
--
Salvatore.
- Original Message -
From: Patrick [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello folks,
I was wondering, talking about VoIP, Asterisk or whatever related to it
What is the Function or Service you really need to create your own business,
simplify service issue, increase your market-cap ?
Is it there but is it not open-source or free ?
I would like collect informations
What is the recommend Digium Card for a PRI in NA ?
I want to interface a Asterisk Server to a Samsung iDCS System, and have
available T1 w/DNIS, or a PRI w/DID, the asterisk server would need to appear
as a Telco provided Circuit.
Slot Availability.
Four PCI-Express Slots x8 (1 full-length/1
On 9/28/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Sep 28, 2007 at 05:28:21PM +0100, Ade Vickers wrote:
Hi folks,
I've been playing around with an Asterisk server in my office for a few
weeks now, and I've got it pretty much nailed down the way I want it,
which
is nice.
I've a big problem with SIP forwarding back into 'ringing groups'
creating what can only be described as call storms :-(
I have a 'ringing groups' of SIP phones with an effective dialplan (much
simplified) like so:
; Purchase ledger
[ptsn_inbound]
exten =
Hi,
Can anyone tell me the pros and cons of Proximity Detection using
bluetooth versus using GSM cell phone with receivers. I like the idea of
calls be transferred to my cell phone when I am away from the office
and I would like to implement such a system.
Thanks
Chuck Bunn
Hi,
Can anyone tell me if the Motorola Q has its Bluetooth always on like
the IPhone? I want to use the Motorola Q in a Proximity Detection setup
like that described on nerdvittles.com. I know the Treo 650 does not
work well since the display must be on for the bluetooth to be on and
this
On Friday 28 September 2007 10:56:19 Rilawich Ango wrote:
Thanks.
Actually, I want to have some information about the call transfer just
like to queue_log in queue.
According to your message, there is no such mechanism to associate the
call in call transfer. How about any variable that I can
Hello,
I am replacing an exisiting call center with a new asterisk based
solution. This will initially consist of to phone servers. The first
being the main PBX, and the second being a predictive dialer. The
dialer will have sip extensions for all the agents, while the main pbx
will hand
Whoops! Forgot to change it for SIP devices.
Of course you need to change your queue member devices to SIP and not
Local/${ARG1} as I've got agents and other complications in mine.
You might need a context or not, see what happens!
Rob
Here is corrected version (I think will work, untested
At 08:01 9/28/2007, Per Jessen wrote:
Douglas Garstang wrote:
Also be sure that you have a very redundant network configuration.
Too often I see people spend a great deal of time and money to get
redundant servers when their switches, firewalls, routers, etc are not
even capable of handling
Tilghman Lesher wrote:
That's true if you use mpg123 for MOH... that's the old way. The recommended
method now is to use native file format, which is saved per channel. So every
channel gets the message started from the beginning.
Aah - cheers for that :) I havnt updated in a while I
Mojo with Horan Company, LLC wrote:
*6 is for *N, for people to remember (N)ight mode. In my *6 extension,
I create a mutex in a sense, the file called 'night_mode' in /home/pbx
-- this lets me determine if night mode is enabled via external systems,
like those written in PHP for a
Greetings,
I know the hardcore guys will laugh, but I put together a quick .nanorc
config for asterisk. I tried to include all the applications listed on
the latest install. Please feel free to send any suggestions/updates my
why. I think this will go a long way to helping out the new guys
I am having an odd audio problem. See setup diagram below. When a call
comes in it get routed through the 1st asterisk box (currently running
1.2) through another asterisk box (running 1.4.11). All audio is good.
When I upgraded the 1st asterisk box to 1.4.11. A call comes in, relays
to the
Ondrej Valousek wrote:
[Sep 20 10:14:32] WARNING[30706]: chan_sip.c:2963 sip_call: No audio
format found to offer. Cancelling call to phone3
Asterisk 1.4 does not have G.722 transcoding, only passthrough support.
It can connect G.722 channels together, and record or playback G.722
audio files,
On 9/27/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Wednesday 26 September 2007 18:39:31 Mark Quitoriano wrote:
Some company asked me to do audits with there asterisk boxes. Is there a
standard that i should be following in auditing? anyway can give me a
start
what to do with asterisk
An example similar to one that exists in many dialplans:
exten = _011.,1,Dial(Zap/g1/${EXTEN})
which would match any international number as dialed from North America
because, depending on what country you'd be calling, the number of
digits after the 011 would differ. As such, putting the
On Sep 28, 2007, at 4:52 PM, Mojo with Horan Company, LLC wrote:
To use the
wildcard characters, 'X', 'N', or '.', I had to also prefix my
extension with '_', which enables pattern matching.
Don't forget you also have Z which if I recall its 1-9, N is 2-9 and
X is 0-9
/b
Peder, I have all the permissions in mysql user. I can query my database
from the local box.
Mik Cheez, yes, it is. mysql.sock is in /var/lib/mysql/
Asterisk and Mysql are in the same PC
I still have the same error and don't know what to do.
help plz!
thanks in advance,
Renzzo
Mik Cheez wrote:
Hi,
is there a way to use g729 in meetme?
Thanks!
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or
On Fri, Sep 28, 2007 at 01:28:18PM -0500, Doug wrote:
How do you do that when your single network connection is gone?
Any suggestions on dual-wan routers? We can't get this
stupid Twin-Wan to work:
http://www.xincom.com/twinwan.php
A PC?
--
Tzafrir Cohen
On 9/26/07, Brian Roy [EMAIL PROTECTED] wrote:
Anyone have a better idea? Or do they have anything like this so I'm not
putting it together?
If its PRI why don't you try:
exten = 00,1,Set(PRI_CAUSE=27)
exten = 00,2,Hangup
Or cause code 17
17 = User Busy. The number dialed
At 20:53 9/28/2007, Tzafrir Cohen wrote:
On Fri, Sep 28, 2007 at 01:28:18PM -0500, Doug wrote:
How do you do that when your single network connection is gone?
Any suggestions on dual-wan routers? We can't get this
stupid Twin-Wan to work:
http://www.xincom.com/twinwan.php
A PC?
Andrew Joakimsen wrote:
On 9/26/07, Brian Roy [EMAIL PROTECTED] wrote:
Anyone have a better idea? Or do they have anything like this so I'm not
putting it together?
If its PRI why don't you try:
exten = 00,1,Set(PRI_CAUSE=27)
exten = 00,2,Hangup
Or cause code
On Fri, Sep 28, 2007 at 03:34:29PM +0200, Philipp Kempgen wrote:
bilal ghayyad wrote:
In the outbound, I read in the documents the Wildcard
match by using the . (period), but I did not
understand how Wildcard will work (like what)?
http://en.wikipedia.org/wiki/Wildcard_character
As
On 9/26/07, Tom Moore [EMAIL PROTECTED] wrote:
If you've got a bandwidth of something that low you'll probably want to use
g723.1 or g729 on this line.
If your lucky you'll be able to place two calls at once over this link.
You won't be able to do anything else though.
Tom
If you really
On 9/28/07, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
Can anyone tell me if the Motorola Q has its Bluetooth always on like
the IPhone? I want to use the Motorola Q in a Proximity Detection setup
like that described on nerdvittles.com. I know the Treo 650 does not
work well since the display
Andrew Joakimsen wrote:
On 9/26/07, Tom Moore [EMAIL PROTECTED] wrote:
If you've got a bandwidth of something that low you'll probably want to use
g723.1 or g729 on this line.
If your lucky you'll be able to place two calls at once over this link.
You won't be able to do anything else
Andrew Joakimsen wrote:
On 9/28/07, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
Can anyone tell me if the Motorola Q has its Bluetooth always on like
the IPhone? I want to use the Motorola Q in a Proximity Detection setup
like that described on nerdvittles.com. I know the Treo 650 does not
52 matches
Mail list logo
