Hi Kai-Uwe,
thank you for your answer. but it doesn't work.
i use this dialplan.
exten = 22,n,Answer()
exten = 22,n,NoCDR()
exten = 22,n,WaitExten(2)
exten = 22,n,Set(CHANNEL(musicclass)=music)
exten = 22,n,Set(CHANNEL(language)=de)
exten = 22,n,Read(roomid,conf-getconfno,6,1)
exten =
On 07/09/10 04:15, C F wrote:
Dial with M option
I don't see how executing a macro BEFORE connecting to the called
channel will help with my issue. Am I missing something?
Thanks,
Barry
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-- Bandwidth and Colocation
Hi,
I am replying on my previous post from last friday.
My pri debugging gives me other statuses as seen at the asterisk application
layer!
In logging below i set the CALLERPRES()=prohib in the dialplan.
When the PRI is used for dialing out the followig is logged:
Calling Number (len=13) [
On Mon, Sep 6, 2010 at 12:32 PM, Barry Miller asterisk-us...@notanet.netwrote:
After upgrading my small test system from Debian Etch-Lenny via a
complete reinstall, I find my g729 hostid has changed. Same machine,
same CPU, same NIC! It doesn't seem reasonable that I have to burn
my one
On Tue, Sep 7, 2010 at 3:11 AM, Daniel Knoll dan...@danielknoll.de wrote:
Hi Kai-Uwe,
thank you for your answer. but it doesn't work.
i use this dialplan.
c exten = 22,n,NoCDR()
exten = 22,n,WaitExten(2)
exten = 22,n,Set(CHANNEL(musicclass)=music)
exten = 22,n,Set(CHANNEL(language)=de)
Hi,
Does Asterisk, provide the option within its voice mail system to be able to do
the following: We want to have an option to be able to have two mail boxes on
one ext. For example: we want to be able to have a French Mailbox and a English
one...
So I can say the following...
You have
Welcome to the right mailinglist :)
Asterisk can definately do that.
All you need to do is make a extension, with an ivr and some sound files.
Have a look at these links:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
Mailboxes dont have to be tied to an extension. The way I would set this
up would be set up the user where his/her primary language mailbox would be
100 and the secondary mailbox would be 101. Then do your dialplan like
this
Exten = 1234,1,playback(3forfrench)
Exten =
Hi Paul,
i set Answer() .. just Cut the first, my fault.
is that the normal case, to treat errors like wrong conference Room?
Daniel
Am 07.09.2010 um 15:01 schrieb Paul Belanger:
On Tue, Sep 7, 2010 at 3:11 AM, Daniel Knoll dan...@danielknoll.de wrote:
Hi Kai-Uwe,
thank you for your
On 09/06/2010 10:28 AM, Olivier wrote:
Hi,
With a 1.4.35 or 1.6.1.19, I'm facing this behaviour :
- extension 7002 is a SIP hard phone currently configured to forward
incoming calls to extension 7003, when a call is unanswered within a 10s
time frame
- when extension 7001 is calling
On 09/07/2010 12:03 AM, Jeff Brower wrote:
Steve-
We are in the process of debugging a voice quality issue for a client of
ours that is a VoIP services provider. The client uses a softphone that
runs on a pjsip stack.
When placing a call using the softphone, it negotiates the use of G729
On Tue, Sep 07, 2010 at 04:02:29AM -0700, Kyle Kienapfel wrote:
On Mon, Sep 6, 2010 at 12:32 PM, Barry Miller
asterisk-us...@notanet.netwrote:
After upgrading my small test system from Debian Etch-Lenny via a
complete reinstall, I find my g729 hostid has changed. Same machine,
same
On Tuesday 07 September 2010 04:29:54 Barry O'Donovan wrote:
On 07/09/10 04:15, C F wrote:
Dial with M option
I don't see how executing a macro BEFORE connecting to the called
channel will help with my issue. Am I missing something?
You're missing that the macro is executed on the CALLED
2010/9/7 Kevin P. Fleming kpflem...@digium.com
On 09/06/2010 10:28 AM, Olivier wrote:
Hi,
With a 1.4.35 or 1.6.1.19, I'm facing this behaviour :
- extension 7002 is a SIP hard phone currently configured to forward
incoming calls to extension 7003, when a call is unanswered within a
Hello list,
is their a way to keep having a dialtone for the calling party when the
macro is executed ?! Or not ?!
Kind regards,
Jonas.
Original Message
Subject:Macro when calling cellphone (GSM) + silence when connecting
Date: Mon, 06 Sep 2010 16:23:46 +0200
Hi!
is their a way to keep having a dialtone for the calling party when the
macro is executed ?! Or not ?!
Consider using either FollowMe() or using the G option of Dial() instead.
Philipp
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Hi,
I have created one SIP extension in Asterisk and configured that extension
in SPA 3102. And connected one FAX machine to the SPA3102 and one to
Asterisk.
The problem is if I try to send FAX from SPA3102 to Asterisk i am not able
to send. But if the same if I try to send from Asterisk to
Yes. Just the lone integrated NIC that's always been there. NO hardware
changes. Still eth0 with the same MAC address.
Do you have any additional, soft network interfaces defined? For
example, have you enabled OpenVPN, and thus loaded either the
tap or tun network-interface drivers? Do you
On Tue, Sep 07, 2010 at 10:58:18AM -0700, Dave Platt wrote:
Note that ifconfig will not necessarily show all of your
interfaces (hard- or soft-) - only the active, configured ones.
ifconfig -a would help here. Kernel upgrades often seem to bring in new
default interfaces.
If this turns out to
I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys
SPA942 SIP phone and outgoing SIP and IAX routes.
When I dial local PSTN numbers from the SPA942 using the FXO channels I
observe a 5-7 second delay between when the PSTN number answers the call
and when Asterisk connects
Hi,
I don't have any g729 codec license. But by reading Barry's complaint I get
to think that it is really unfair that Digium can't renew his license or
something.
I am a Debian user myself and I understand the need to upgrade from etch to
lenny (and to squeeze in no time).
Having a kernel built
Hi All;
I would like to use Asterisk for a call center, but really does not know if
Asterisk support the following in a good way:
1) Ability to do an inteligent routing, so to route the call to the proper
skill group based on the caller information?
2) If I can create skill groups and then
On 09/07/2010 06:40 PM, Philipp von Klitzing wrote:
is their a way to keep having a dialtone for the calling party when the
macro is executed ?! Or not ?!
Consider using either FollowMe() or using the G option of Dial() instead.
Philipp
So with this G option I send the caller and
I've lost licenses because I'd modified my redondant dual-ethernet cards into
two separate interfaces. I've bought the Digium G729 card even though my
server could handle the load with software compression just to get rid of this
potential issue. I'm quite happy not having to think about this.
I'm having a wierd problem. Somewhere around 1-2% of the time, the
first DTMF digit dialed gets dropped. This is occurring during a
SpeechBackground application call. If the caller reenters the digits
when given a second chance, all is OK.
Any suggestions how to debug this intermittent
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank
Tarczynski
Subject: [asterisk-users] 5-7 second connection delay in outgoing FXO calls
I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner
Subject: [asterisk-users] Losing first DTMF digit (with ASR)
I'm having a wierd problem. Somewhere around 1-2% of the time, the
first DTMF digit
On Tue, Sep 07, 2010 at 07:43:45PM +0100, Tiago Geada wrote:
Hi,
I don't have any g729 codec license. But by reading Barry's complaint I get
to think that it is really unfair that Digium can't renew his license or
something.
I am a Debian user myself and I understand the need to upgrade
Is the message played very long/short? I play a lot of my speechbackground
messages with beep in front (speechbackground(beepfoo)) so my user doesn't
start hitting DTMF until the message starts playing.
It's about six seconds. I've seen the problem myself and I'm dialing the
first DTMF digit
On Tue, 7 Sep 2010, Tiago Geada wrote:
Hi,
I don't have any g729 codec license. But by reading Barry's complaint I get
to think that it is really unfair that Digium can't renew his license or
something.
I am a Debian user myself and I understand the need to upgrade from etch to
lenny (and
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner
Subject: Re: [asterisk-users] Losing first DTMF digit (with ASR)
Is the message played very long/short? I play a lot of my
speechbackground
On 09/07/2010 02:16 PM, Barry Miller wrote:
I posted here before contacting Digium. They have been helpful.
Here is what I've found :
An old Etch dmesg shows this:
eth0: VIA Rhine II at 0x1cc00, 00:13:d4:f5:e3:e6, IRQ 10.
eth0: MII PHY found at address 1, status 0x786d advertising
Richard
Who is the carrier that the calls are flowing in from?
Bruamt
From: ken...@gnat.com (Richard Kenner)
I'm having a wierd problem. Somewhere around 1-2% of the time, the
first DTMF digit dialed gets dropped. This is occurring during a
1. Vestec, Lumenvox or other?
Vestec
2. How many digits of DTMF are you aiming for (using SPEECH_DTMF_MAXLEN?)
6
3. Are you presenting DTMF back (verbose ${SPEECH_TEXT(0)}) ?
Similar. There's a NoOp that display what was originally that value in
the log.
--
Is there a way to solve the mess on CDR caused by
CDR Transfer? anyway, by paid support, no paid, or another way...
Im going crazy about this. My boss is really furious because he
dont understand nothing on the CDR.
I tried the 1.6.2.11, Asterisk 1.8
Who is the carrier that the calls are flowing in from?
It's a Paetec PRI into an NEC SV8300, then QSIG from there to Asterisk.
--
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New to Asterisk? Join us
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner
Sent: Tuesday, September 07, 2010 2:54 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Losing first DTMF digit (with ASR)
Who is the carrier that the
I have seen simalar issues with some audiocodes gear. I adjusted the early
media options on the pri in the audiocodes gateway and that fixed my
issues. We have also seen this when calls come in from Level 3 toll free
some times their gatways screw with things. We found adding an manual
answer
So it's a PRI/DAHDI connection.
Yeah, but with switchtype=qsig, though that difference isn't likely
relevant here.
Is SpeechBackground the first item in the context?
No. There are plenty of others, starting with an Answer(200). Then
a whole bunch of Speech* applications to load grammar
El 07/09/10 14:49, Fabiano Carlos Heringer escribió:
Is there a way to solve the mess on CDR caused by CDR Transfer?
anyway, by paid support, no paid, or another way... Im going crazy
about this. My boss is really furious because he don´t understand
nothing on the CDR.
I tried the 1.6.2.11,
Just for grins, do this command
/bin/grep num sent /var/log/VestecASRE/Port-10500_2010-09-07.log
This should show you all of the DTMF processed by the grammars today.
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Just for grins, do this command
/bin/grep num sent /var/log/VestecASRE/Port-10500_2010-09-07.log
This should show you all of the DTMF processed by the grammars today.
It doesn't show any. Isn't DTMF processed by Asterisk and not the ASR?
Anyway, I can now reproduce this in a simpler case:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner
Sent: Tuesday, September 07, 2010 4:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Losing first DTMF digit (with ASR)
if you use SpeechBackground, DTMF is under ASR control (returned in
SPEECH_TEXT(0) ).
It is returned in SPEED_TEXT(0), but it's still being done by Asterisk,
not the ASR engine.
Anyway, your other test indicates that the DTMF press
used to stop the prompt is being eaten by the ASR or
On Tue, Sep 7, 2010 at 2:56 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Appreciate your kindly advise and help.
With the proper dial-plans, testing and additional development,
everything you have listed is possible.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
On 09/08/2010 03:23 AM, Gordon Henderson wrote:
On Tue, 7 Sep 2010, Tiago Geada wrote:
Hi,
I don't have any g729 codec license. But by reading Barry's complaint I get
to think that it is really unfair that Digium can't renew his license or
something.
I am a Debian user myself and I
On Mon, Sep 06, 2010 at 03:32:35PM -0400, Barry Miller wrote:
After upgrading my small test system from Debian Etch-Lenny via a
complete reinstall, I find my g729 hostid has changed. Same machine,
same CPU, same NIC! It doesn't seem reasonable that I have to burn
my one no-hassle
Em 07/09/2010 17:15, Miguel Molina escreveu:
El 07/09/10 14:49, Fabiano Carlos Heringer escribi:
Is there a way to solve the mess on CDR
caused by
CDR Transfer? anyway, by paid support, no paid, or another
way...
From the way you posted your questions I understood that you want to
give the option for the person being called to enter DTMF then hangup
and keep the calling person alive (continue in dial plan) the M option
will do that for you.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
On Tue, Sep
Wow thats a LONG signature.
Oh and BTW, the future of this world, how many years in the future?
On Tue, Sep 7, 2010 at 9:16 AM, Frenette, Rob
rob.frene...@bullyingcanada.ca wrote:
Hi,
Does Asterisk, provide the option within its voice mail system to be able
to do the following: We want to
On 9/7/2010 9:05 PM, asterisk-users-requ...@lists.digium.com wrote:
Subject: [asterisk-users] 5-7 second connection delay in outgoing FXO calls
I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys
SPA942 SIP phone and outgoing SIP and IAX routes.
When I dial local PSTN
On 3/09/10 8:32 AM, Arnaldo Giacomitti Junior wrote:
There´s a way to get the channel signalling in dialplan?
I have changed the code in channels/chan_dahdi.c and includes:
Upload it as a patch to the issue tracker:
http://issues.asterisk.org
--
Cheers,
Matt Riddell
Hello!
I want to get rtcp stats to cdr. (btw, I run asterisk 1.6.2.11)
There is howto here http://www.voip-info.org/wiki/view/Asterisk+RTCP
But I (and my users) do bridged calls from dahdi to sip, so in h
extension channel is dahdi , and it doesn't contain rtcp stats.
There is info about
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