Stefan Schmidt s...@sil.at writes:
if i understand you right you have one server (peer) where thousands of
devices are connected and every device is registered to asterisk, and so
every options packet will come from asterisk to this device, right?
If you have a sip routing server like ser,
Hello,
How can I do bitwise AND operations on a variable? I want to check the bits
set in the HANGUPCAUSE, but can't find a way to do it.
--
Alex Hermann
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asterisk-users
Give this a whirl:
${MATH(HANGUPCAUSEbit)}
Got it here:
https://issues.asterisk.org/view.php?id=9891
Alex Hermann wrote:
Hello,
How can I do bitwise AND operations on a variable? I want to check the bits
set in the HANGUPCAUSE, but can't find a way to do it.
--
Alex Balashov
Sorry, didn't carefully read my own link. :-)
I guess it should be:
${MATH(HANGUPCAUSE AND bit)}
Alex Balashov wrote:
Give this a whirl:
${MATH(HANGUPCAUSEbit)}
Got it here:
https://issues.asterisk.org/view.php?id=9891
Alex Hermann wrote:
Hello,
How can I do bitwise
Hi,
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
Thanks,
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Peder wrote:
Decent product, but their support and development are horrible. I showed
them that their SIP over TCP implementation was broken and their reply was
use udp
Such a shame it sounds like it has gone down hill, previously when I've
spoken to them the standard response was that
Thomas Kenyon schrieb:
Peder wrote:
Decent product, but their support and development are horrible. I showed
them that their SIP over TCP implementation was broken and their reply was
use udp
Such a shame it sounds like it has gone down hill, previously when I've
spoken to them the
Steve Underwood schrieb:
Klaus Darilion wrote:
Atis Lezdins schrieb:
On Mon, Jun 8, 2009 at 2:06 PM, Klaus
Darilionklaus.mailingli...@pernau.at wrote:
Hi!
I have the following problem with Asterisk 1.4.23:
ATA w/ T.38 Asterisk ATA w/o T.38
Dear all,
I connected a normal phone line to the FXO port but the call is not being
processed. The following is the output to asterisk console when I dial 9150
9 is the prefix I configured and 150 is a local service in to know the
current time
*CLI -- Executing Dial(SIP/-d365, Zap/1/150)
Benny Amorsen schrieb:
Klaus Darilion klaus.mailingli...@pernau.at writes:
Asterisk does not forward the 488 back to the caller, but hangs up the
callee's call leg. Further, the caller's call leg will not be hung up.
Is somebody aware of this problem and a fix?
This should be fixed in
make sure to set canreinivte=yes for both peers
regards
klaus
Mindaugas Kezys schrieb:
Hello,
I want to send Media outside Asterisk server, e.g. between peers.
In CLI I see:
· [Jun 8 13:13:58] VERBOSE[19112] logger.c: -- Native bridging
SIP/5060-b7dc5218 and
Benny Amorsen schrieb:
Stefan Schmidt s...@sil.at writes:
if i understand you right you have one server (peer) where thousands of
devices are connected and every device is registered to asterisk, and so
every options packet will come from asterisk to this device, right?
If you have a sip
Louis-David Mitterrand schrieb:
Hi,
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
Digium's BRI cards are also based on Cologne Chip - thus you could try
Digiums BRI drivers.
On Tue, Jun 09, 2009 at 02:45:26PM +0200, Klaus Darilion wrote:
Louis-David Mitterrand schrieb:
Hi,
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
Digium's BRI cards are also based on Cologne Chip - thus you could try
Digiums BRI
On Tue, Jun 09, 2009 at 02:57:00PM +0200, Louis-David Mitterrand wrote:
On Tue, Jun 09, 2009 at 02:45:26PM +0200, Klaus Darilion wrote:
Louis-David Mitterrand schrieb:
Hi,
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
On Tue, Jun 09, 2009 at 02:45:26PM +0200, Klaus Darilion wrote:
Louis-David Mitterrand schrieb:
Hi,
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
mISDN 1.1? mISDN2 (chan_lcr)? chan_dahdi?
Digium's BRI cards are also based on
Klaus Darilion schrieb:
Louis-David Mitterrand schrieb:
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
Digium's BRI cards are also based on Cologne Chip - thus you could try
Digiums BRI drivers.
Interview with Kevin Fleming about Zaptel,
Has anyone set it up so that an inside call and an outside call get
different unavailable messages?
j
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Here is one way you could do it. Have your inside calls be busy and outside
be unavailable. Then use CALLERID(num) to determine the source of the call
then do Voicemail(blah,b) or Voicemail(blah,u) depending on the call source.
Here's an example using the ex-girlfriend logic; 100 is
Jeff LaCoursiere schrieb:
Has anyone set it up so that an inside call and an outside call get
different unavailable messages?
Yes.
AGI() -- optional
if(...){...}else{...}
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan
Since you restated this clearly, it seems to be a pretty simple 2 place
patch; app_voicemail.c would need the option added to play the new messages
(add an I and O option to the existing b,u,s,g and j options) and
app_voicemailmain.c would need the option added to record them. Steve
Edwards eats
On Tue, 9 Jun 2009, Philipp Kempgen wrote:
Jeff LaCoursiere schrieb:
Has anyone set it up so that an inside call and an outside call get
different unavailable messages?
Yes.
AGI() -- optional
if(...){...}else{...}
I appreciate the sarcasm - I suppose I should have been a bit more
On Tue, Jun 9, 2009 at 12:31 AM, Steve Reposcmu...@gmail.com wrote:
What is the magic to compile chan_dahdi.so in asterisk 1.6.1.x? I'm
on centos 5.3.
Also, asterisk 1.4.25 cannot compile chan_dahdi as well while the
previous versions do. What changed or what am i missing?
There probably
Jeff LaCoursiere schrieb:
Has anyone patched the voicemail app such that inside/outside messages
are CLEARLY supported, i.e. they have menu options for recording inside
and outside greetings, and the app can accept some form of argument
specifying an inside or outside call?
You could
I try festival and espeak, festival is better then espeak but I need more
natural sounds
On Mon, Jun 8, 2009 at 4:57 PM, equis software equissoftw...@gmail.comwrote:
I need to imlplement an IVR service where customers call and put a
telephone number, then I reproduce the name and address.
On Tue, Jun 09, 2009 at 04:04:29PM +0300, Tzafrir Cohen wrote:
On Tue, Jun 09, 2009 at 02:45:26PM +0200, Klaus Darilion wrote:
Louis-David Mitterrand schrieb:
Hi,
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
mISDN
My .02 - It is really easy to miss/mess up the supporting files when
upgrading. Make sure you are using a segregated tree to compile from.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Moises Silva
Sent:
Klaus Darilion klaus.mailingli...@pernau.at writes:
;timerb=32000 ; Call setup timer. If a provisional response is not
received
; in this amount of time, the call will autocongest
; Defaults to 64*timert1
Thanks! Will try that. Just what I was looking
On Tue, Jun 09, 2009 at 04:50:39PM +0200, Louis-David Mitterrand wrote:
Does the digium driver also work for Beronet's (or Junghanns) 4BRI
and 8BRI cards?
https://issues.asterisk.org/view.php?id=13897
Tested to properly work with the OpenVox cards. Likely to work well but
not operate LEDs
On Tue, 2009-06-09 at 14:04 +, Jeff LaCoursiere wrote:
Has anyone patched the voicemail app such that inside/outside messages
are CLEARLY supported, i.e. they have menu options for recording inside
and outside greetings, and the app can accept some form of argument
specifying an inside
On Tuesday 09 June 2009 10:32:01 Jared Smith wrote:
On Tue, 2009-06-09 at 14:04 +, Jeff LaCoursiere wrote:
Has anyone patched the voicemail app such that inside/outside messages
are CLEARLY supported, i.e. they have menu options for recording inside
and outside greetings, and the app
On Tue, Jun 9, 2009 at 4:50 PM, Philipp
Kempgenphilipp.kemp...@amooma.de wrote:
Thomas Kenyon schrieb:
Peder wrote:
Decent product, but their support and development are horrible. I showed
them that their SIP over TCP implementation was broken and their reply was
use udp
Such a shame
On Tue, 9 Jun 2009, Steve Repo wrote:
On Tue, Jun 9, 2009 at 4:50 PM, Philipp
Kempgenphilipp.kemp...@amooma.de wrote:
Thomas Kenyon schrieb:
Peder wrote:
Decent product, but their support and development are horrible. I showed
them that their SIP over TCP implementation was broken and
I sent this message again, the first one got stopped for moderation
because it was barely over 40k.
We had another dahdi problem this morning and got a dahdi_test -v. When
calling into Span 2 it was just dead air. There was also an HDLC Abort
at the start of the problem this time as well, but
On Tue, 2009-06-09 at 12:11 -0500, Tilghman Lesher wrote:
It does, but it also makes listening to messages rather difficult, as the
fallback for languages only works in one direction.
That's a very valid point... My intention was to be able to set a
language for just the sound prompts, not for
Just founnd a weirdy. My end is Asterisk 1.2.32 using an IAX2 link to
the US.
The IP address of the remote end changed (though in the config file it's
registered as a name i.e. asterisk.remote.end), my system didn't
recognised the IP change, it must be cached once and then the cached
value used
Did you do an IAX2 show peer on it?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Kennedy
Sent: Tuesday, June 09, 2009 1:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] IAX2 issue?
On Tue, Jun 09, 2009 at 02:02:50PM -0500, Danny Nicholas wrote:
Did you do an IAX2 show peer on it?
Remote end unreachable and old IP address
Steve
--
NetTek Ltd UK mob +44 7775 755503
UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455
I got the call pick up to work with the Digium AA50 and the GXP2000.
Here is what I used in the dial plan:
exten=_*8.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten=_*8.,n,Pickup(${EXTEN:2...@pickupmark)
You dial *8 and the extension that is ringing and you will intercept the call.
Hope that
What does your iax2 show registry look like?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Kennedy
Sent: Tuesday, June 09, 2009 2:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
On Tue, 2009-06-09 at 19:58 +0100, Steve Kennedy wrote:
Just founnd a weirdy. My end is Asterisk 1.2.32 using an IAX2 link to
the US.
The IP address of the remote end changed (though in the config file it's
registered as a name i.e. asterisk.remote.end), my system didn't
recognised the IP
On Tue, 9 Jun 2009, Steve Kennedy wrote:
Just founnd a weirdy. My end is Asterisk 1.2.32 using an IAX2 link to
the US.
The IP address of the remote end changed (though in the config file it's
registered as a name i.e. asterisk.remote.end), my system didn't
recognised the IP change, it must
On Tue, 9 Jun 2009, Jeff LaCoursiere wrote:
I now have a handful of the new 3140 video phones, and I have to admit I am
pretty impressed. I have a handful of the 3000 phones, and was somewhat
impressed. We are planning to offer a service around the 3140 soon, and I am
trudging through
Hello,
I'm running into an issue with TLS transport and I am probably missing
something obvious.
We are trying to configure an extension to use TLS for the transport.
The extension can make outbound calls using TLS, but inbound calls fail.
The extension configuration in sip.conf is set to
At 06:20 6/9/2009, Philipp Kempgen wrote:
Thomas Kenyon schrieb:
Peder wrote:
Decent product, but their support and development are horrible. I showed
them that their SIP over TCP implementation was broken and their reply was
use udp
Such a shame it sounds like it has gone down
Doug wrote:
Linksys looks good in comparison.
I've found (in the past) linksys support to be quite good (although not
used much of their voice products). Certainly they are in my experience
much better than the kit they sell.
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Hi Steve;
Currently with Dahdi, for the below configuration that we were using it in the
rc.local, how it will be?
I think there should be something new to be used instead of ztcfg, what it is?
And what about other lines? They need to be changed?
touch /var/lock/subsys/local
/sbin/modprobe
Hit another problem in my tutorial in converting over from Cisco
CallManager to Asterisk.
I have been following the instructions at :
http://voip-info.capatres.com/wiki/view/Asterisk+Cisco+CallManager+Integ
ration.html on intergrating Asterisk and Cisco CallManager.
I can make calls from CCM to
Make sure you are stripping the 8 on inbound calls to that H323 gateway
under CCM and that it has a valid search space to find your extensions...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Ezell
Sent: Tuesday, June 09, 2009
Christian,
Dial through is fine for me and it it worked really well. The
problem was that I wastesting it with a Soft Phone that would not
support DTMF. Once I replaced the phone, it worked.
Cheers
Manoj
-Original Message-
From: asterisk-users-boun...@lists.digium.com
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