Hi all,
I'm experiencing some problems with i4l and i can't find a solution. I'm
using
Eicon Diva 1 BRI
Eicon Diva Server 4 Bri
A ISDN PBX where I connect the first ISDN card (exten 204 in the
ISDN PBX) and a ISDN phone (exten 210 in the ISDN PBX)
Suse 8.1
Hi,
I had the same problem when i tried i4l, and as far as I remember the
solution was to set
the outgoing msn to the msn of the isdn-line.
From my old modem.conf:
incomingmsn=*
outgoingmsn=123456,123457
device = /dev/ttyI0
device = /dev/ttyI1
best regards,
Nils
On Sat, 1
PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Hatzis,
Michael
Enviado el: martes, 14 de diciembre de 2004 23:21
Para: Andrew Furey; Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: RE: [Asterisk-Users] Busy message on ISDN cards?
I had the same problem even though it was with capi
PROTECTED] On Behalf Of Andrew
Furey
Sent: Tuesday, 14 December 2004 4:43 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Busy message on ISDN cards?
Hi all,
I'm new to asterisk and not too knowledgeable on ISDN, so please be
gentle :)
I have a dual-channel Eicon Diehl Diva card in a Debian
Hi all,
I'm new to asterisk and not too knowledgeable on ISDN, so please be gentle :)
I have a dual-channel Eicon Diehl Diva card in a Debian Woody box with
kernel 2.4.27, connecting to a Telstra (Australia) Onramp Home Highway
ISDN line. I'm pretty certain the card and line both work since
On Tue, 2004-06-22 at 21:44, Aaron J. Angel wrote:
After doing some quick research, it appears HANGUPCAUSE is only implemented
in chan_zap and chan_sip. What about the other channels?
They are out of luck until someone creates a patch to add that feature
to the other channels. I belive
There are other users running the latest CVS-HEAD reporting that problem
(asterisk segfaults when unable to create channel). Maybe you have to
revert to a previous version till the bug is fixed. ( cvs -D )
OK, thanks, will try that (btw, cvs -D is an invalid command)
Have you any idea why
Keith Waters schrieb:
...
I configured:
exten = _[123456789],1,NoOp(.call for .${EXTEN})
I consider a short key for [1-9] at least
as useful as N for [2-9], maybe even more useful.
For my internal purposes I'm using E for [1-9].
Am I the only one, who is missing something short
for [1-9]?
Hi!
I think the whole idea of busy or unavailable is flawed.
Asterisk sets ${CAUSECODE} with the cause of the call being
cleared. You can use this to determine what you want to do.
For exmaple if the cause code indicates unallocated then
you should give the caller some indication that
Hi!
I consider a short key for [1-9] at least
as useful as N for [2-9], maybe even more useful.
For my internal purposes I'm using E for [1-9].
Am I the only one, who is missing something short
for [1-9]?
Certainly not! :-) Create a [request] entry at bugs.digium.com
Cheers, Philipp
...
Certainly not! :-) Create a [request] entry at bugs.digium.com
Done.
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Philipp von Klitzing [EMAIL PROTECTED] wrote:
I consider a short key for [1-9] at least as useful as N for [2-9],
maybe even more useful.
For my internal purposes I'm using E for [1-9].
Am I the only one, who is missing something short for [1-9]?
Certainly not! :-) Create a [request]
Hi,
-Original Message-
I consider a short key for [1-9] at least as useful as N for [2-9],
maybe even more useful.
For my internal purposes I'm using E for [1-9].
Am I the only one, who is missing something short for [1-9]?
Certainly not! :-) Create a [request] entry
: [Asterisk-Users] Busy message
Eric Wieling [EMAIL PROTECTED] wrote:
On Tue, 2004-06-22 at 18:43, Simon Brown wrote:
This should be listed as a bug - it is not logical to go to busy,
when in fact the extension is unavailable.
I think the whole idea of busy or unavailable is flawed.
Asterisk
Aaron J. Angel schrieb:
...
Does Z not work?
...
Yes, it does. I had to look in the most recent docs first.
Sorry!
Roger.
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Hi Keith
Keith Waters wrote:
There are other users running the latest CVS-HEAD reporting that problem
(asterisk segfaults when unable to create channel). Maybe you have to
revert to a previous version till the bug is fixed. ( cvs -D )
OK, thanks, will try that (btw, cvs -D is an invalid command)
I don't think it's documented, but Z specifies 1-9
exten = _Z,1,NoOp(.call for .${EXTEN})
On Wed, 2004-06-23 at 03:52, Roger Schreiter wrote:
Keith Waters schrieb:
...
I configured:
exten = _[123456789],1,NoOp(.call for .${EXTEN})
I consider a short key for [1-9] at least
as
That message is created by the Voicemail application. Check your
extensions.conf and see what your action is for when the call can not
be connected.
For example, a correct dialplan for a SIP extension would read:
exten = _200Z,1,Dial(SIP/${EXTEN},20)
exten = _200Z,2,Voicemail(u${EXTEN})
exten =
For example, a correct dialplan for a SIP extension would read:
exten = _200Z,1,Dial(SIP/${EXTEN},20)
exten = _200Z,2,Voicemail(u${EXTEN})
exten = _200Z,102,Voicemail(b${EXTEN})
exten = _200Z,103,Hangup
Hi All... I'm a newbie, just busy getting to grips with asterisk.
I've set up the
Hi Keith,
Hi All... I'm a newbie, just busy getting to grips with asterisk.
I've set up the following, but it causes a segfault when I call somebody who
is offline:
exten = _[123456789],1,NoOp(.call for .${EXTEN})
exten = _[123456789],2,Dial(SIP/${EXTEN},60,tr)
exten =
Are you running Redhat or Fedora? If so, read this thread for a solution:
http://lists.digium.com/pipermail/asterisk-users/2004-January/031953.html
Nope, SUSE SLES 8
regards,
Keith
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Keith Waters wrote:
Are you running Redhat or Fedora? If so, read this thread for a solution:
http://lists.digium.com/pipermail/asterisk-users/2004-January/031953.html
Nope, SUSE SLES 8
There are other users running the latest CVS-HEAD reporting that problem
(asterisk segfaults when unable to
On Mon, 2004-06-21 at 23:26, Simon Brown wrote:
When I dial a SIP phone which is specified in the sip.conf, but the phone is
not connected, Asterisk gives the message The user at Extension XXX is on
the phone
Shouldn't the message be the unavailable message?
Is there something wrong with
PROTECTED]
Subject: Re: [Asterisk-Users] Busy message
On Mon, 2004-06-21 at 23:26, Simon Brown wrote:
When I dial a SIP phone which is specified in the sip.conf, but the
phone is not connected, Asterisk gives the message The user at
Extension XXX is on the phone
Shouldn't the message
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Wednesday, 23 June 2004 0:47
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Busy message
On Mon, 2004-06-21 at 23:26, Simon Brown wrote:
When I dial a SIP phone which is specified in the sip.conf
: [Asterisk-Users] Busy message
*I* think it should go to unavailable, but it has always gone to busy.
On Tue, 2004-06-22 at 16:34, Simon Brown wrote:
Then shouldn't Asterisk be changed so it jumps to unavailable in the
dial plan? Surely this would be the correct way of working.
Simon Brown
. Dial will continue with the next priority
on time-out (which generally happens when the device isn't answered).
-Original Message-
From: Simon Brown [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 22, 2004 6:44 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Busy message
On Tue, 2004-06-22 at 18:43, Simon Brown wrote:
This should be listed as a bug - it is not logical to go to busy, when in
fact the extension is unavailable.
I think the whole idea of busy or unavailable is flawed. Asterisk
sets ${CAUSECODE} with the cause of the call being cleared. You can
Eric Wieling [EMAIL PROTECTED] wrote:
On Tue, 2004-06-22 at 18:43, Simon Brown wrote:
This should be listed as a bug - it is not logical to go to busy,
when in fact the extension is unavailable.
I think the whole idea of busy or unavailable is flawed.
Asterisk sets ${CAUSECODE} with the
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Wednesday, 23 June 2004 8:43
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Busy message
*I* think it should go to unavailable, but it has always gone to busy.
On Tue, 2004-06-22 at 16:34, Simon Brown wrote
Brown [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 22, 2004 6:44 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Busy message
This should be listed as a bug - it is not logical to go to
busy, when in fact the extension is unavailable.
Simon
-Original Message
On Tue, 2004-06-22 at 19:31, Aaron J. Angel wrote:
What would the contents of CAUSECODE be when set? I can't find
documentation of this anywhere.
Sorry, it's ${HANGUPCAUSE} Asterisk hangup cause as documented in
docs/README.variables. The cause code listing can be found in
Logged in bugtracker as Bug #1893
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, 23 June 2004 11:37
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Busy message
The issue has been suggested several times
Eric Wieling [EMAIL PROTECTED] wrote:
On Tue, 2004-06-22 at 19:31, Aaron J. Angel wrote:
What would the contents of CAUSECODE be when set? I can't find
documentation of this anywhere.
Sorry, it's ${HANGUPCAUSE} Asterisk hangup cause as
documented in docs/README.variables. The cause code
[EMAIL PROTECTED] wrote:
Sorry, it's ${HANGUPCAUSE} Asterisk hangup cause as
documented in docs/README.variables. The cause code listing
can be found in include/asterisk/cause.h
As twisted points out: Hangup cause is different than why it couldn't
create the channel. If that's the case, why
When I dial a SIP phone which is specified in the sip.conf, but the phone is
not connected, Asterisk gives the message The user at Extension XXX is on
the phone
Shouldn't the message be the unavailable message?
Is there something wrong with my set up or is this a bug with Asterisk?
Simon
Why not have dial just dial, then have applications like WaitForAnswer,
WaitForDisconnect etc...?
This would give more granularity to the call flow control and allow
someone to get brave and write a WaitForHuman or whatever.
Hmm... I can't think of too many instances where the
Why not have dial just dial, then have applications like WaitForAnswer,
WaitForDisconnect etc...?
This would give more granularity to the call flow control and allow
someone to get brave and write a WaitForHuman or whatever.
Hmm... I can't think of too many instances where the
There's not really a way to do that that right now, although we could add
something like AST_CONTROL_INUSE which could represent that the channel is
in use actually. Wouldn't be extremely difficult to do, but would INUSE
and BUSY be the same? If not, where do we jump to?
Mark
On Wed, 11 Jun
Hmm... this gets quickly back to my long-standing desire to have more
comprehensive call completion codes being handed back by the channels
to the dialplan.
The current method of throwing certain replies into a big bucket
called Busy and others into a big bucket called Error and
auto-jumping
On Fri, 2003-06-13 at 16:07, John Todd wrote:
Hmm... this gets quickly back to my long-standing desire to have more
comprehensive call completion codes being handed back by the channels
to the dialplan.
Just a couple of comments.
I agree with jtodd about the call completion codes, but I'd
Is it possible to have both a busy and an away message when the call
waiting feature is enabled?
extensions.conf
...
exten=403,1,Dial,Zap/3|10
exten=403,2,Voicemail2,u403
exten=403,103,Voicemail2,b403
...
Because I have enabled call waiting, I can't see how it will be possible
to get the busy
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