In my opinion the dialplan isn't where that logic belongs.
/b
On Oct 3, 2007, at 12:32 AM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED]
wrote:
Hello,
I see that most people are using the extensions.conf syntax (most
of the
examples and questions here use that syntax). recently
but they do, apparently
On 10/2/07, robert boardman [EMAIL PROTECTED] wrote:
Hi All
I'm having problems posting to this list, no bounces the mails just
dont show
any advice how to get the postings through is there filtering?
robb
___
On 10/2/07, Ken Williams [EMAIL PROTECTED] wrote:
Any suggestions would be greatly appreciated.
Try removing all the echo cancel stuff just to see if that makes any
difference at all.
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--Bandwidth and Colocation Provided by
Hi
i'm setting up a hylafax server, using iaxmodem to talk with asterisk
(asterisk and hylafax are both on the same lan).
Can i setup on the same host (Hylafax) multiple iax accounts ? (each
account is used by a iaxmodem instance).
The account can be on the same port or should i change the port
Where would you suggest all the logic goes Brian?
Garth
Garth van Sittert
BSc (Physics Computer Science)
-
Main: 08600 BITCO
Phone: +27 (0)11 875 6900
Fax:+27 (0)11 875 6901
Mobile: +27 (0)83 791 6662
Email: [EMAIL PROTECTED]
MSN:[EMAIL PROTECTED]
Web:
On 10/3/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Tuesday 02 October 2007 16:55:52 Brian West wrote:
On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote:
anyway still if there's a hack for meetme to work with g729 codec
this won't be an issue. So is there a hack or patch that i can
I need to select a line from the Zap group channel
using the SIP Phone (not FXO and not FXS ports).
ignorepat does not work?
Also, what is the method to let the second dial tone
has another tone frequency?
Regards
Bilal
No, ignorepat is for FXS ports (FXS ports use FXO
On Tuesday 02 October 2007 19:30:44 Thomas Kenyon wrote:
Is there an advantage to having a Queue members URI in the form:
SIP/User (or indeed IAX2/User)
Over
Local/number@context
?
I know that the latter will allow you to do things like set counting
logic etc. through dialplan
Mee too, a lot of the messages I'm sending seem to disappear.
l.
In data Tue, 02 Oct 2007 22:38:26 +0200, robert boardman
[EMAIL PROTECTED] ha scritto:
Hi All
I'm having problems posting to this list, no bounces the mails just
dont show
any advice how to get the postings through is
I believe that using the Local/[EMAIL PROTECTED] format will give you a bit
more
flexibility in the dialplan design, as there is an added degree of
indirection. In the end I think this is only marginally costier than the
raw channel format (unless you use the /n option) and should provide
Mark,
Or, in other words, you cannot mix compressed data. You must first
decompress the data for mixing, then recompress it for transmission.
yeah i still don't understand. this is what i want to do. I want
asterisk not to compress and decompress codecs. so either i can use SLIN
as my
In article [EMAIL PROTECTED],
Mark Quitoriano [EMAIL PROTECTED] wrote:
yeah i still don't understand. this is what i want to do. I want asterisk
not to compress and decompress codecs. so either i can use SLIN as my codec
for my SIP or IAX. or i can remove SLIN codec in meetme and change it
Tilghman Lesher wrote:
Or, in other words, you cannot mix compressed data. You must first
decompress the data for mixing, then recompress it for transmission.
During both operations, there is a potential for signal degradation.
Ummm, why?? Unless you can explain some technical reason
Hi list,
Running Asterisk 1.4.10:
When using the M() option for Dial to execute a macro, then executing a
Read within the macro, once streaming of the audio file specified in
Read has completed, and the channel attempts to read input from the
destination channel where the macro is executed, the
Same here
lenz wrote:
Mee too, a lot of the messages I'm sending seem to disappear.
l.
In data Tue, 02 Oct 2007 22:38:26 +0200, robert boardman
[EMAIL PROTECTED] ha scritto:
Hi All
I'm having problems posting to this list, no bounces the mails just
dont show
any advice how to
Peter Fern wrote:
Tilghman Lesher wrote:
Or, in other words, you cannot mix compressed data. You must first
decompress the data for mixing, then recompress it for transmission.
During both operations, there is a potential for signal degradation.
Ummm, why?? Unless you can explain
Hi:
I installed A102d sangoma's card successfully but Asterisk doesn't answer to
incoming call from pstn and console doesn't show any message of incoming call
in the other word when I diall the number of E1 I can't connect to asterisk and
dial the number of extension.
I'd apreciateany idea.
If I upgrade libpri 1.4.0 to 1.4.1, do I then need to recompile
asterisk even though I'm not upgrading asterisk?
--
-
Chris Stinson
Network Operations Center
ISDN-Net, Inc.
615-221-4200 x103
[EMAIL PROTECTED]
___
--Bandwidth and Colocation
On 3 Oct 2007, at 10:16, Mark Quitoriano wrote:
On 10/3/07, Tilghman Lesher [EMAIL PROTECTED]
wrote: On Tuesday 02 October 2007 16:55:52 Brian West wrote:
On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote:
anyway still if there's a hack for meetme to work with g729 codec
this won't
On Wed, Oct 03, 2007 at 07:16:17AM -0500, Chris Stinson wrote:
If I upgrade libpri 1.4.0 to 1.4.1, do I then need to recompile
asterisk even though I'm not upgrading asterisk?
To the best of my knowledge: no. Unless you have some non-standard
patches to one of the versions (and not to the
For me this is very common. As soon as you define a problem in words
(Email or otherwise) instead of concepts in your head, boom, the answer
jumps out.
That's why I really like whiteboards. I can draw the concept and then
put it into words, side by side. Almost always figure out my issue
Ken Williams wrote:
Below is a copy of my log, zapata.conf extensions.conf that relate to
the ZAP lines. Basically when we dial out it takes on 10-12 seconds
before the ZAP line actaully picks up. I'm hoping to find out what the
cause is for this as it's causing user grief with extremely
Don Pobanz wrote:
the Asterisk release contains a large number of
bug fixes for all parts of Asterisk.
I am thankful to see the amount of fixes that have gone into this
release. However, seeing this many fixes does not give me a warm fuzzy
feeling that we won't see a lot more fixes in
must be blacklisted, i have posted like 4 messages and none are showing up.
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi all,
The thing that has bugged me about Asterisk since I first started
playing with it, is the fact that the pound sign/hash/octothorp doesn't
resolve digit conflicts or cancel timing on a variable length string such
as a tie line code or when you call numbers in a country whose
On Wednesday 03 October 2007 15:41:08 William F. Acker WB2FLW +1-303-722-7209
wrote:
Hi all,
The thing that has bugged me about Asterisk since I first started
playing with it, is the fact that the pound sign/hash/octothorp doesn't
resolve digit conflicts or cancel timing on a variable
fateme fatah wrote:
Hi:
I installed A102d sangoma's card successfully but Asterisk doesn't
answer to incoming call from pstn and console doesn't show any message
of incoming call in the other word when I diall the number of E1 I
can't connect to asterisk and dial the number of extension.
On Wed, 2007-10-03 at 14:14 +0300, Zoa wrote:
Same here
Yes, I'm aware that some people are having problems posting to the
mailing list, and I'm working with Digium's IT staff to try to correct
the problems. It seems to be related to our inbound spam filtering.
(The weird thing is that new
You have various scripting languages things like that can go in!
/b
On Oct 3, 2007, at 4:12 AM, Garth van Sittert wrote:
Where would you suggest all the logic goes Brian?
Garth
Garth van Sittert
BSc (Physics Computer Science)
-
Main: 08600 BITCO
Phone: +27 (0)11 875
On Wednesday 03 October 2007 06:09:01 Peter Fern wrote:
Tilghman Lesher wrote:
Or, in other words, you cannot mix compressed data. You must first
decompress the data for mixing, then recompress it for transmission.
During both operations, there is a potential for signal degradation.
Yehavi Bourvine +972-8-9489444 wrote:
Hello,
I see that most people are using the extensions.conf syntax (most of the
examples and questions here use that syntax). recently I've translated all my
dial plan to AEL syntax and I find it much easier, especially when you need
IFs.
Why
On Tue, Oct 02, 2007 at 06:20:54PM +0200, Artifex Maximus wrote:
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 12:47:55PM +0200, Artifex Maximus wrote:
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 10:06:06AM +0200, Artifex
Lee Jenkins wrote:
Yehavi Bourvine +972-8-9489444 wrote:
Hello,
I see that most people are using the extensions.conf syntax (most of the
examples and questions here use that syntax). recently I've translated all my
dial plan to AEL syntax and I find it much easier, especially when
But I believe Cisco is the only manufacturer producing a phone with a
gigabit port for connecting a desktop pc. Anyone know of any other?
Glenn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Monday, October 01, 2007 7:18 PM
To: Asterisk
Wouldnt that take a very large portion of datapower, to startup the parsers and
such, instead of having the whole dialplan natively in Asterisk.
We always try to do as much as possible in dialplan, so that we are not reliant
on external scripts.
Kind Regards
Jon Leren Schøpzinsky
Pong
On 10/2/07, Steve Totaro [EMAIL PROTECTED] wrote:
must be blacklisted, i have posted like 4 messages and none are showing up.
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asterisk-users mailing list
To UNSUBSCRIBE
Let us not forget that AEL cannot be stored in a database therefore
rendering you unable to utilize realtime.
AEL converted into standard extensions.conf syntax in the dialplan.
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So, I updated to 1.4.12 last night and it appears my problem is mostly
gone now. Not sure what the difference was, but it now takes about 3
seconds before the ZAP line picks it up. I was on 1.4.10.1 before that,
and yes POTS. Removing the echo cancellation at this point makes no
difference, not
Hello all,
Is it possible to store, read and write configuration files in an SQLite3
database instead of using the configuration files inside the /etc/asterisk/
directory? If it is then can you point me to the right documentation on how
to do this or probably hints on how to do this?
Thank you
I can't help you with that. I only wanted to point out that ignoreopat
is not what you need.
On Polycom SIP phones you continue dialtone by placing a , in the
phone's dialplan. SIP phones have their own internal dialplan that is
not part of Asterisk's dialplan. You would have to check the
On Oct 3, 2007, at 9:39 AM, Jon Schøpzinsky wrote:
Wouldnt that take a very large portion of datapower, to startup the
parsers and such, instead of having the whole dialplan natively in
Asterisk.
We always try to do as much as possible in dialplan, so that we are
not reliant on external
On 10/3/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 06:20:54PM +0200, Artifex Maximus wrote:
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 12:47:55PM +0200, Artifex Maximus wrote:
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Hello All,
For some odd reasons my Asterisk is keep on loosing registration of my
SIP devices. On the SIP device it shows I am RESISTED but when I do sip
show peers it shows my sip endpoints are UNREACHABLE. And it keeps on
flapping Peer '903456' is now UNREACHABLE! and Peer '903456'
I have been following this discussion. You do have a point. However, the
way * works right now. If a channel does not require trans-coding to get
into a conference, coder usage is counted. So I really do not know what
difference putting the transcoding in meetme is going to make. I mean,
how could
But his preference of G729 is to save bandwidth.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton
Sent: Wednesday, October 03, 2007 8:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] meetme
Doug,
Look at the list. It seems you and Nitesh Divecha may be having the
same problem. Maybe you guys can confirm that you have the same issue
and figure out what is in common, such as Asterisk version or whatever.
Thanks,
Steve
Doug Reid wrote:
Hi Steve
I have tried a constant ping and
If bandwidth were not an issue, I would think everyone would opt for
ulaw or alaw. Why compress and use CPU cycles and G729 licenses if
there were no bandwidth issues?
Thanks,
Steve totaro
Wai Wu wrote:
But his preference of G729 is to save bandwidth.
-Original Message-
From:
Lee Jenkins wrote:
Why most people don't use it? Am I missing something?
I think it looks too much like C.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
Hi Nitesh,
The reachable/unreachable determination is not connected to registration
expiry parameters in any way.
There is a qualify= parameter (that has a default value, and I think it
may be on by default) that is associated with all SIP peers. It is
basically a way to say that the SIP
For another tone frequency for the outside dialtone, try putting this
value [EMAIL PROTECTED],[EMAIL PROTECTED],[EMAIL
PROTECTED];*(.4/0/1),10(*/0/2+3) in the Outside
Dialtone field. It will give you a slight pause followed by a different
dialtone frequency. On a Linksys/Siprua 941, that would
Looks like a bug they have fixed with the latest 1.4.x release.
Please, can we have a 1.2.x spoon? Instead of just security fixes, the
spoon should also include bug fixes and backports or new functionality
in later Asterisk versions.
Thanks,
Steve Totaro
Ken Williams wrote:
So, I updated
Eric ManxPower Wieling wrote:
Let us not forget that AEL cannot be stored in a database therefore
rendering you unable to utilize realtime.
AEL converted into standard extensions.conf syntax in the dialplan.
Doesn't this render having used AEL pointless?
--
Thank you and have a
Its just a different way to express the same thing in a more fluid way.
/b
On Oct 3, 2007, at 10:33 AM, Anthony Francis wrote:
Doesn't this render having used AEL pointless?
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To each his own. I like the flat files personally, they are more fluid
to me.
Thanks,
Steve
Brian West wrote:
Its just a different way to express the same thing in a more fluid way.
/b
On Oct 3, 2007, at 10:33 AM, Anthony Francis wrote:
Doesn't this render having used AEL pointless?
I'm growing fond of XML.
/b
On Oct 3, 2007, at 10:39 AM, Steve Totaro wrote:
To each his own. I like the flat files personally, they are more
fluid
to me.
Thanks,
Steve
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I have a client who wants a Meetme box with 12 FXO ports, to connect
to Analogue lines coming from an Ericsson PBX.
It looks like I could do this with four different hardware configurations:
a) three TDM04B cards (based on TDM400P)
b) one TDM04B and one TDM808B
c) one TDM804B (or TDM854B?) and
Now on to another problem that we've had as far as I know since the
beginning of using Asterisk 9+ months ago. I've been trying very hard
to knock this problem out but regardless of what I do, it's still there.
So, the problem is, when a call is in the parking lot, it then times out
after
GNUbie wrote:
Hello all,
Is it possible to store, read and write configuration files in an
SQLite3 database instead of using the configuration files inside the
/etc/asterisk/ directory? If it is then can you point me to the right
documentation on how to do this or probably hints on how
Tony Mountifield wrote:
I have a client who wants a Meetme box with 12 FXO ports, to connect
to Analogue lines coming from an Ericsson PBX.
It looks like I could do this with four different hardware configurations:
a) three TDM04B cards (based on TDM400P)
b) one TDM04B and one TDM808B
c)
Tony Mountifield wrote:
I have a client who wants a Meetme box with 12 FXO ports, to connect
to Analogue lines coming from an Ericsson PBX.
It looks like I could do this with four different hardware configurations:
a) three TDM04B cards (based on TDM400P)
b) one TDM04B and one TDM808B
c)
Those are all analog though, aren't they? What about a channel bank into
a digital card? Might that be cheaper than shelling out for 12 FXO
ports and the cards to hold them?
Just wanted to throw that out there before the discussion started :)
Tony Mountifield wrote:
I have a client who
None are great options. I'd use a T1 card and a channel bank.
At minimum I'd do the single 2400P. IRQ problems are going to be a
bear with multiple cards.
-Darren
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tony
On Wed, 2007-10-03 at 09:33 -0600, Anthony Francis wrote:
Eric ManxPower Wieling wrote:
Let us not forget that AEL cannot be stored in a database therefore
rendering you unable to utilize realtime.
AEL converted into standard extensions.conf syntax in the dialplan.
Thomas Kenyon wrote:
Tony Mountifield wrote:
I have a client who wants a Meetme box with 12 FXO ports, to connect
to Analogue lines coming from an Ericsson PBX.
It looks like I could do this with four different hardware configurations:
a) three TDM04B cards (based on TDM400P)
b) one TDM04B
Mojo with Horan Company, LLC wrote:
Those are all analog though, aren't they? What about a channel bank into
a digital card? Might that be cheaper than shelling out for 12 FXO
ports and the cards to hold them?
Just wanted to throw that out there before the discussion started :)
It
Thank you very much, Mark. =)
On 10/4/07, Mark Michelson [EMAIL PROTECTED] wrote:
GNUbie wrote:
Hello all,
Is it possible to store, read and write configuration files in an
SQLite3 database instead of using the configuration files inside the
/etc/asterisk/ directory? If it is then
I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard
TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards.
Each group of T1's have the primary D on 24 and the secondary D on 96.
The first server (ts20) and the last server (ts22) can playback
demo-congrats fine. The
Ken Williams wrote:
Now on to another problem that we've had as far as I know since the
beginning of using Asterisk 9+ months ago. I've been trying very hard
to knock this problem out but regardless of what I do, it's still there.
[from-internal]
include = parkedcalls
I have
On Wed, 3 Oct 2007, Steve Edwards wrote:
I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard
TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards.
Each group of T1's have the primary D on 24 and the secondary D on 96.
The first server (ts20) and the last server
I have an asterisk process that is consuming over 100mb (according to
top). Show channels says 167 active channels and 53 active calls.
It's an old install -- 1.2.7.1, but it has custom code that needs to be
updated before moving to a more recent release.
I'm assuming that 100mb is indicative
On Wednesday 03 October 2007 20:48:37 Steve Edwards wrote:
install -- 1.2.7.1, but it has custom code that needs to be
updated before moving to a more recent release.
I'm assuming that 100mb is indicative of a memory leak (probably in my
code).
How can I get a dump (preferably without
Steve Totaro wrote:
must be blacklisted, i have posted like 4 messages and none are showing up.
That's what I thought, too, but there's some weirdness going on with
Digium's list server spam filtering.
Anyway, you'll probably see this one :)
-Stephen-
Steve Edwards wrote:
I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard
TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards.
Each group of T1's have the primary D on 24 and the secondary D on 96.
The first server (ts20) and the last server (ts22) can playback
Greetings,
I have a dialplan that calls the dictate application, but I want to do
some post-processing on the RAW file created. The post processing is
working fine as long as the dictation application exits gracefully, but
fails when the user simply hangs up.
How can I make sure the system()
Steve Edwards wrote:
On Wed, 3 Oct 2007, Steve Edwards wrote:
I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard
TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards.
Each group of T1's have the primary D on 24 and the secondary D on 96.
The first server
Jim Canfield wrote:
Greetings,
I have a dialplan that calls the dictate application, but I want to do
some post-processing on the RAW file created. The post processing is
working fine as long as the dictation application exits gracefully, but
fails when the user simply hangs up.
How
Have you tried adding an 'h' extension in addition? If the caller hangs
up in the middle of priority 1 of extension 123, it should then jump to
priority 1 of extension h and continue.
;Add to the test-dictation context:
exten = h,1,System(post_processing_script.sh)
OR
;Not tested, but maybe
Mojo with Horan Company, LLC wrote:
Have you tried adding an 'h' extension in addition? If the caller hangs
up in the middle of priority 1 of extension 123, it should then jump to
priority 1 of extension h and continue.
Thanks,
That works perfectly.
Someone who's having trouble posting to the list should try placing
[asterisk-users] or Re:in the subject line of a new email they
send (near the END of the subject so it doesn't obscure the actual
subject or have superfluous Re:'s near the beginning) to see if the
spam filter is more
In features.conf, I have uncommented the transfer features under feature
map, but I still cannot transfer using a POTS phone on an IAXy adapter.
I think I am missing something here Any help is appreciated.
Here is features.conf:
;
; Sample Parking configuration
;
[general]
It would be ugly, but you could prefix a zap channel or group number
before the phone number to dial. Using groups for an example:
exten = _*X*X.,1,Dial(ZAP/g${EXTEN:1:1}/${EXTEN:3})
exten = _*XX*X.,1,Dial(ZAP/g${EXTEN:1:2}/${EXTEN:4})
so dialing *4*18005551212 dials out over zap group 4...
On Wednesday 03 October 2007 22:21:24 Michael Munger wrote:
In features.conf, I have uncommented the transfer features under feature
map, but I still cannot transfer using a POTS phone on an IAXy adapter.
I think I am missing something here Any help is appreciated.
Do you have t and/or T
Michael Munger wrote:
In features.conf, I have uncommented the transfer features under feature
map, but I still cannot transfer using a POTS phone on an IAXy adapter.
I think I am missing something here…. Any help is appreciated.
Those features are triggered via DTMF, not using a
On Wed, 3 Oct 2007, Steve Totaro wrote:
Steve Edwards wrote:
I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard
TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards.
Each group of T1's have the primary D on 24 and the secondary D on 96.
The first server
So what, then, is the procedure to transfer a call from a POTS phone on
the FXS port of an IAXy?
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, October 03, 2007
Steve Edwards wrote:
[trunkgroups]
trunkgroup = 1,24,96
spanmap = 1,1,0
spanmap = 2,1,2
spanmap = 3,1,3
spanmap = 4,1,1
You
Can you describe exactly what you lose by using the dynamic queue member
alternative? We tried to ensure that no functionality was lost in this
transition, so if there is something that was missed please let us know
what it is and we'll try to take care of it.
Now, i'm finally trying to
When I was unable to figure out the IAXy's methods, I went with
Asterisk's features.conf -- ## for blindxfer, and never looked back.
That worked quite well.
Michael Munger wrote:
So what, then, is the procedure to transfer a call from a POTS phone on
the FXS port of an IAXy?
Yours,
Hi list,
Is there a limit on the length of an extension? I have an 18 byte long
extension, when issuing goto, Asterisk comes back with invalid
extension on the console. Anyone had this experience before?
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If you want this to work nicely dont settle for anything else than a
channel bank
On 10/3/07, Thomas Kenyon [EMAIL PROTECTED] wrote:
Mojo with Horan Company, LLC wrote:
Those are all analog though, aren't they? What about a channel bank into
a digital card? Might that be cheaper than
I am assuming you mean 18 digits long. it shouldnt be a problem you
mind posting your configs?
On 10/3/07, Wai Wu [EMAIL PROTECTED] wrote:
Hi list,
Is there a limit on the length of an extension? I have an 18 byte long
extension, when issuing goto, Asterisk comes back with invalid
extension
It just dawned on me, that I can just press the hook button momentarily
to open up a second IAX channel, dial the number, and hangup to complete
the transfer.
Thanks everyone!
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
Steve Edwards wrote:
On Wed, 3 Oct 2007, Steve Totaro wrote:
Steve Edwards wrote:
I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard
TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards.
Each group of T1's have the primary D on 24 and the secondary D
Kevin P. Fleming wrote:
Steve Edwards wrote:
[trunkgroups]
trunkgroup = 1,24,96
spanmap = 1,1,0
spanmap = 2,1,2
spanmap = 3,1,3
spanmap
Hi list,
I recently purchased an ISDNguard from Junghanns. It came with no
software and there is no sign on their website or in any of their
documentation where to download it. I have looked in
http://www.junghanns.net/downloads/ and there is no sign of it there
either. The only thing remotly
I am trying to use PHP to reload the extensions in an Asterisk
installation. I keep getting this error:
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
when I run the script by visiting the URL; however, if I run the script
from the command line, it runs just fine (works
Here is how i overcome this problem,
ignorpat = 9
exten = 9*,1,Dial(ZAP/1/w)
press 9* from your handset and after 1 second you have POTS line dial tone
on your phone,
On 10/3/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
It would be ugly, but you could prefix a zap channel or
On Wed, 3 Oct 2007 08:35:06 -0500, Tilghman Lesher wrote:
I invite you to try it. You could make a lot of really smart people look like
fools if you're able to mix compressed audio together without decompressing,
or you might make yourself look like a fool, because you get back garbage for
Michael Munger wrote:
I am trying to use PHP to reload the extensions in an Asterisk
installation. I keep getting this error:
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
when I run the script by visiting the URL; however, if I run the script
from the command
If you are running the script from a web server, the script gets
executed with the web server process permissions, hence, probably does
not have access to /var/run/asterisk.ctl.
You can give permissions to your web server, or better yet, dont
execute the command using shell_exec, better open a
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