hello all,
how isit possible to give a busy signal if the line is in use?
For me it is ringing and signaling on my phone, when i call out and i
get another call.
My hint is this:
exten = 31,hint,SIP/1000131SIP/1000131a
i have one softphone an one hardphone
Regards René
--
René Enskat
Jerry wrote:
Hi Dovid,
I am actually now working on massproducing door
openers that will work with asterisk. It will have an
rj45 port and then a port to plug the door opener in
to. Please contact me off list if you are interested.
This is an old message, but I was wondering if you are still
Ciao Carlos,
I finished to try to install Asterisk Now 1.4.0 on an AMD 3800 dual
processor machine.
The install lookups on the search for the Sata drive, since however
it loads the sata_sil driver it doesn't work.
I have had some problems with Asterisk Now, until I switched to text
mode
Am Donnerstag, den 21.12.2006, 11:28 +0800 schrieb Rilawich Ango:
How about:
exten = _X.,1,Answer
Does it include all numbers and characters?
As of the docs, no. It should only match 0123456789
See
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
BR
Anselm
Klaverstyn, David C wrote:
There seems to be something in Asterisk that disconnects the call at 1 hour.
At 59 minutes there is a beep then 1 minute later the call is dropped.
I have a basic install Asterisk Ver. 1.2.13. I have not specifically
said that calls are to be disconnected
Am Mittwoch, den 20.12.2006, 14:42 -0500 schrieb Doug Crompton:
Anthony,
Ok I understand. The 011 is unique though and I guess the problem is
the length of the remaining digits. This could vary based on country?? and
I suspect there is no unique rule that could be applied??? I have not
RL == Richard Lyman [EMAIL PROTECTED] writes:
RL grr, i hate when i typo (and reply to my own posts) exten =
RL s/,2,Set(CALLERID(name)=OUTSIDE NAME|CALLERID(num)=xx)
Heh, if you want to chase typos, perhaps you should add an underscore
before ?
/Benny
Carlos Chavez wrote:
CAS signalling on span 1 conflicts with HDLC with FCS check on channel
My guess is not to use HDLC, as the error says above, that it conflicts
with CAS.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
Rilawich Ango wrote:
Any command to refresh or clear the whole ast database?
asterisk -rx 'stop now'
rm astdb
asterisk
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
On Thu, 21 Dec 2006 08:33:27 -0500, Doug Lytle wrote
Carlos Chavez wrote:
CAS signalling on span 1 conflicts with HDLC with FCS check on channel
My guess is not to use HDLC, as the error says above, that it
conflicts with CAS.
I wish it were that easy and obvious. I only found
Greetings,
Currently my asterisk box is using Voicepulse. It works fine with the
exception that people need to enter the 1+area code for local calls.
I'd like to get around this if possible. The following is what I have
in my extensions.conf..
exten =
Is there a way I can create a _NXX extension and insert 1 and areacode
when dialing?
exten = _NXX,1,Set(CALLERID(num)=6162997590)
exten = _NXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/1514${EXTEN})
replace 514 with your area code
hth
___
Hi Phil,
Using your example:
exten = _NXXNXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/1${EXTEN})
... Would match NXX-NXX- and pop a one in place of what you dialed.
Alex
On 12/21/06, Phil Finkler [EMAIL PROTECTED] wrote:
Greetings,
Currently my asterisk box is using
Phil,
Yeah, I just realized that I didn't answer your question. Time Bandit did
though, look at his solution!
Alex
On 12/21/06, Alex Robar [EMAIL PROTECTED] wrote:
Hi Phil,
Using your example:
exten = _NXXNXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/1${EXTEN})
... Would match
I was playing with aelparse last night and I thought it would be nice if
the output of the it's operation was a little more structured.
I've written a app that allows me to edit ael/conf files from a windows
environment and upload them to the asterisk box, commit a reload,
restart, etc,
On Thu, Dec 21, 2006 at 09:57:50AM -0500, Lee Jenkins wrote:
I was playing with aelparse last night and I thought it would be nice if
the output of the it's operation was a little more structured.
I've written a app that allows me to edit ael/conf files from a windows
environment and
; Dial wether long distance is preceeded by 1 or not
; Dial LD via gizmo
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = _1NXXNXX,2,Macro(failann,${DIALSTATUS})
exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = _NXXNXX,2,Macro(failann,${DIALSTATUS})
One way audio is almost always caused by firewalls / NAT translation. Since
there is neither on IPKall, my guess would be to look at the other end. With
20k + users, most have succeeded in correcting this problem via their
hardware / software. I encourage you to look at the user forum for some
I too am wondering if someone has a contact at Thomson, some of the softkeys
need to either be fixed or have the option to remove (like FwdVM and
Pickup keys).
In addition, has anyone notice a humming noise when using the handset? I can
hear it and so can the person that I am calling.
On
Tzafrir Cohen wrote:
Maybe an optional different file descriptor rather than a dump file?
Would that have been of more use to you?
That could certainly work.
--
Warm Regards,
Lee
___
--Bandwidth and Colocation provided by Easynews.com --
Benny Amorsen wrote:
RL == Richard Lyman [EMAIL PROTECTED] writes:
RL grr, i hate when i typo (and reply to my own posts) exten =
RL s/,2,Set(CALLERID(name)=OUTSIDE NAME|CALLERID(num)=xx)
Heh, if you want to chase typos, perhaps you should add an underscore
before
I think its rPath Linux, based on redhat. I've had some problems with
Asterisk Now. My X100P card was not recognized since it didnt show in the
zap channels in the GUI thats why I switched back to debian and install
Asterisk from source.
On 12/20/06, Carlos Alperin [EMAIL PROTECTED] wrote:
callgroups pickupgroups greater than 31 are not working for sip calls
with 1.2.14 tarball. Anyone know which branches support 64?
John
___
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To UNSUBSCRIBE or update
I'm no C programmer, but is this 32 limit just an array definition somewhere?
Wouldn't it be a no brainer to track it down and increase it so some very large
number?
-Original Message-
From: John Harragin [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 21, 2006 11:56 AM
To:
our * crashed twice in a month with segmentation fault
a core dump. here's the stack trace:
#0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6
#1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6
#2 0xb7e17090 in strdup () from /lib/tls/libc.so.6
#3 0x08057ada in ast_verbose (fmt=0x0) at
Yes thats the bottom line, its mostly the country code which can be 1-3
digits long. There is no rules based solution for this. Historicaly each
country picked a number out of a hat except the US (which had to be
number 1) because as we all know it's the centre of the universe. The
former USSR had
-Original Message-
From: Richard Lyman [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with,
dialplan
Douglas Garstang wrote:
Hello,
We are having a hard time making the GXP-2000 work reliably with Asterisk.
We have several clients using the GXP-2000. These phones are behing NAT
and our Asterisk server has a public IP (no NAT).
The biggest problem we face is the clients complain of random, but
frequent, calls (in or
FYI, astribanks all come with outputs that can be used for door
openers, combined with this product from Vikingelectronics.com that
plugs into any fxs port you should have a complete solution for a
door:
http://www.vikingelectronics.com/products/view_product.php?pid=99
They (viking) has a door
On Thu, 2006-12-21 at 12:07 -0700, Douglas Garstang wrote:
I'm no C programmer, but is this 32 limit just an array definition somewhere?
Wouldn't it be a no brainer to track it down and increase it so some very
large number?
I think pickupgroup is defined as 'unsigned int' somewhere in
Douglas Garstang wrote:
-Original Message-
From: Richard Lyman [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with,
dialplan
Douglas
Has anyone used either the 8 port or 4 port FXO device from
Grandstream? (GXW-4108 or 4104).
They seem to be the lowest cost multi port FXO devices that I can
find, so I'm getting ready to buy the 8 port version. I just want to
see if there are any opinions on the device before I commit to
Eric Jacksch wrote:
You might also want to look at what the legal situation is in your
jurisdiction. Here one only needs the consent of one party to the call,
so I don’t have to advise the callee that the call is recorded if the
caller consents to the recording.
If you are in the U.S.,
I would be very interested in getting an 8 port FXO myself. They are very
new so I don't think there are any used ones out there yet.
Does anybody out there in Canada stock them yet?
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Has anyone used either the 8 port or 4
Greetings folks.
I seem to be having a problem where calls made from an IAX device (three
single-line phones attached to IAXys) do not play the ring tone when
calling out. There's a dial tone when I pick up the phone, and the call
goes through just fine, it just doesn't ring. All my SIP
Andrew Joakimsen ha scritto:
I too am wondering if someone has a contact at Thomson, some of the
softkeys need to either be fixed or have the option to remove (like
FwdVM and Pickup keys).
In addition, has anyone notice a humming noise when using the handset?
I can hear it and so can the
Chris,
These devices are still very new to the market. Finding reviews on
them may be tough still. From our experience its a good little device
for the dollar; but, keep in mind, it's still a low cost gateway and
that normally means don't expect too much.
We've sold few cases here and
Does anyone know the maximum number of
digits for an international phone number?
Doing some searching, it looks like 16
numbers including the 011 is the
maximum number, because 17 is just not
found:
OK:1234567890123456
http://www.google.com/search?q=011X
On 21/12/06, Doug [EMAIL PROTECTED] wrote:
Does anyone know the maximum number of
digits for an international phone number?
Doing some searching, it looks like 16
numbers including the 011 is the
maximum number, because 17 is just not
found:
OK:1234567890123456
On 16:03, Thu 21 Dec 06, Jay Moore wrote:
Greetings folks.
I seem to be having a problem where calls made from an IAX device (three
single-line phones attached to IAXys) do not play the ring tone when
calling out. There's a dial tone when I pick up the phone, and the call
goes through
Hi,
I'm using Asterisk (1.2.13) on Centos 4.4 x86_64 with a TDM2400E for analog
trunks ( extensions) plus some Polycom 501 601 phones.
I have a problem in that the audio via the Polycoms is gated or muted during
quiet parts of the other person's speech.
This results in the start of words being
On 22:56, Thu 21 Dec 06, Peter Bowyer wrote:
On 21/12/06, Doug [EMAIL PROTECTED] wrote:
Does anyone know the maximum number of
digits for an international phone number?
Doing some searching, it looks like 16
numbers including the 011 is the
maximum number, because 17 is just not
found:
you mean we need to remove astdb manual? Totally restart asterisk
even the whole server doesn't do the removement?
On 12/21/06, Doug Lytle [EMAIL PROTECTED] wrote:
Rilawich Ango wrote:
Any command to refresh or clear the whole ast database?
asterisk -rx 'stop now'
rm astdb
asterisk
--
Ben
Thanks. I got it.
On 12/21/06, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Donnerstag, den 21.12.2006, 11:28 +0800 schrieb Rilawich Ango:
How about:
exten = _X.,1,Answer
Does it include all numbers and characters?
As of the docs, no. It should only match 0123456789
See
Hi all, as good?
I try to install asterisk-1.2.14, zaptel-1.2.12,libpri-1.2.4,addons-1.2.5 ,
sounds-1.2.1 and wanpipe-2.3.4-3 and hwec-utils-beta4-2.3.4
But it is not compiling drivers of the Sangoma, why udev's for board in
/dev/zap(1-31, channel,ctl,pseudo,timer) is not created. But when I
I second that. I'm quite happy with the IPKall.com did number I use today.
Only once in the last year was it unavailable when I needed it. So, not
bulletproof, but good enough for me to use all day when I work at home.
On 12/21/06, www.IPKall.com [EMAIL PROTECTED] wrote:
One way audio is
Hi All.
Forgive me, but mine motherboard is ASUS P5GPL-X SE
Thank's
Best Regards
Josue
2006/12/22, Josué Conti [EMAIL PROTECTED]:
Hi all, as good?
I try to install asterisk-1.2.14, zaptel-1.2.12,libpri-1.2.4,addons-1.2.5,
sounds-1.2.1 and wanpipe-2.3.4-3 and hwec-utils-beta4-2.3.4
But it is
We have an application for Asterisk that will require connecting 144 fax
ports into the system. Faxes will route externally over a PRI. The 144
ports are for local fax machines within the building. Not all will be
faxing simultaneously. We just need to be able to provide ports in the
building
stay away from foip stick with channel banks
On 12/21/06, Allen Casteran [EMAIL PROTECTED] wrote:
We have an application for Asterisk that will require connecting 144 fax
ports into the system. Faxes will route externally over a PRI. The 144
ports are for local fax machines within the building.
Question... What is the purpose of the + before the number? Does anyone
actually have to enter it? If so how would you do it? It is not used in
the US but do I see it come in on SIP lines CID. I assume the CID ignores
it in the number as I do not see it on the display. It is however stored
in
Hello the list,
You can use FXS and em signalling to reverse the line polarity temporary to
trigger an external door opener interface.
This is very easy.
Good Luck !
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
I noticed that asterisk will keep the phone record in astdb when the
phone (especially hardphone) unplugged.
After unplug the phone, I still get the phone information in astdb:
database showkey SIP/Registry/1234
/SIP/Registry/1234 :
10.14.43.31:40876:60:1234:sip:[EMAIL
I think the + convention started off because different countries have
different international access codes. Well, on GSM networks, + can be a
part of the number to represent the international access code ( the
traditional access code in India is 00 for international). So to call
Digium, from my
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