Re: [asterisk-users] Reduce the wav file size
Thanks for reply,
This recording is meetme conference recording. normally meetme file is can
wav format. is there any ways to change the meetme file into gsm format.
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
On Wed, Oct 5, 2011 at 11:48 AM, Kevin P. Fleming wrote:
> On 10/05/2011 01:30 AM, mahesh katta wrote:
>
>> Hi list,
>>
>> How to reduce the meetme wav file size in asterisk. how can I
>> automatically reduce this file size.
>>
>> exten =>
>> _8600[1-2]XX,1(record),SetVar(**MEETME_RECORDINGFILE=/var/**
>> spool/asterisk/meetme/**conference_recording-${EPOCH}-**
>> ${USER}-${TIMESTAMP}-${EXTEN})**;
>> exten => _8600[1-2]XX,2,Meetme,${EXTEN}**|Fr
>>
>
> A WAV file is (generally) uncompressed audio; it's size cannot be reduced
> except through compression, which would need to be reversed in order to play
> back the file.
>
> If you want to store recordings using less disk space, store in a
> compressed format like GSM, G.729 or something else supported in Asterisk.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
> --
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>
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[asterisk-users] Questions on Dahdi
I have naive question. I do not have any hardware on my asterisk host. All I have are either SIP trunk for DID or hardware ATA which bridges the asterisk to PSTN. Do I need Dahdi install? Do i have ztdummy for timing issue? I encounter problem in this when I try to install Dahdi latest but I found it is not running, Instead it runs when service starts but I can't find its status when I type in service dahdi status. I am using Asterisk 1.8.7 on centos 5.7 32 bit. CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay before ringing from PSTN`s call
Hi,
I commented the option callerid in the file dahdi-channels.conf without
success, My SIP phone still ring after 4-5 secondes :-(
; Span 1: WCTDM/0 "Wildcard TDM410P" (MASTER)
;;; line="1 WCTDM/0/0 FXSKS (EC: MG2 - INACTIVE)"
signalling=fxs_ks
;callerid=asreceived
I am living in Canada, so I guess I should use USA signaling ? If so in
which file ?
Message: 7
Date: Tue, 04 Oct 2011 14:49:55 -0400
From: John Novack
Subject: Re: [asterisk-users] Delay before ringing from PSTN`s call
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: neo haux
Message-ID: <4e8b5553.2030...@stromberg-carlson.org>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"
You need to make sure your (DAHDI or ZAPTEL ) is set up properly for your
country's CLID protocol
In the US CLID is sent between the first and second rings, and with a proper
configuration Asterisk waits a ring before processing the call
Other parts of the world use different methods and protocols
You will need to dig into that first.
John Novack
neo haux wrote:
> Hi
>
> I am testing a degium TDP400P (2fxo+2fxs) on my asterisk
>
> I configured incoming calls from pstn to ring my SIP phone (extension :
100)
>
> cat extensions.conf
> ...
> [from-pstn]
> exten => s,1,Dial(SIP/100,10)
> same => n,VoiceMail(100,u)
>
>
>
>
> root@PC-debian:/etc/asterisk# cat dahdi-channels.conf
> ...
> ...
> ...
> ;;; line="1 WCTDM/0/0 FXSKS (EC: MG2 - INACTIVE)"
> signalling=fxs_ks
> callerid=asreceived
> group=0
> context=from-pstn
> channel => 1
> callerid=
> group=
> context=default
> ...
> ...
> ...
>
> What I don`t understand is why the SIPphone rings after 3 secondes after
Astererisk detects the incoming call. Moreover, after hanging off the
external caller the SIPphone continue to ring for 3 seconds.
>
> I did those modifications in the file /etc/asterisk/chan_dahdi.conf
without improuvement ( After restarting Asterisk)
>
> [channels]
> cidstart=ring
> immediate=yes
> faxdetect=no
> usecallerid=no
>
>
> Here is the debug from Asterisk console
>
> *CLI> -- Starting simple switch on 'DAHDI/1-1'
> -- Executing [s@from-pstn:1] Dial("DAHDI/1-1", "SIP/100,10") in new
stack
> == Using SIP RTP CoS mark 5
> -- Called SIP/100
> -- SIP/100-0001 is ringing
> == Spawn extension (from-pstn, s, 1) exited non-zero on 'DAHDI/1-1'
> -- Hanging up on 'DAHDI/1-1'
> -- Hungup 'DAHDI/1-1'
>
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Re: [asterisk-users] call pickup
Am 05.10.2011 20:42, schrieb Marek Cervenka: hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? You can have that with subscriptions/hints, for example Snom phones can display not only a call to one of the peers but also the caller and callee identification. This works jaw to cheek with BLF (busy lamp field) which allows to monitor other extensions' status (in_use, ringing...). Of course you can be member of a pickup group without "monitoring" the status of any of the peers, and you can monitor a peer's status without being in the same pickup group (although not pickup the call then, obviously :-) Regards Martin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 Manager Perl Script Problem [SOLVED]
On Mon, Oct 3, 2011 at 5:01 PM, JR Richardson wrote:
> Hi All,
>
> Trying to upgrade some call servers, in the lab making sure all my
> applications work, ran into an issue with some manager perl scripts
> that pull and reset database info, it seems the command and result
> responses have changed but I'm not sure how to resolve. My scripts
> are using CPAN Asterisk::Manager; and are working fine on asterisk
> 1.2.32 but not on Asterisk 1.8.6.0.
>
> Here is the abbreviated script where 1.2.32 is astman1 and 1.8.6.0 is astman2:
>
>
> #!/usr/bin/perl -w
> use strict;
> use warnings;
> use Getopt::Long;
> use Asterisk::Manager;
>
> ##setup manager connections##
> my $astman1 = new Asterisk::Manager;
> $astman1->user('username');
> $astman1->secret('password');
> $astman1->host('10.10.14.101');
> $astman1->connect || die $astman1->error . "\n";
>
> my $astman2 = new Asterisk::Manager;
> $astman2->user('username');
> $astman2->secret('password');
> $astman2->host('10.10.14.102');
> $astman2->connect || die $astman2->error . "\n";
>
> ##query databases for cnam count##
> $astman1->sendcommand(Action => 'DBGet', Family => 'cnam', Key => 'count');
> my @result1 = $astman1->sendcommand(Event => 'DBGetResponse');
> my $cnamcount1 = "0$result1[7]";
>
> $astman2->sendcommand(Action => 'DBGet', Family => 'cnam', Key => 'count');
> my @result2 = $astman2->sendcommand(Event => 'DBGetResponse');
> my $cnamcount2 = "0$result2[7]";
>
> ##total cnam count##
> my $cnamtotal = ($cnamcount1+$cnamcount2);
>
> ##reset cnam count to 0##
> $astman1->sendcommand(Action => 'DBPut', Family => 'cnam', Key =>
> 'count', Val => '0');
> my @result101 = $astman1->sendcommand(Event => 'DBGetResponse');
> my $cnamreset1 = $result101[1];
>
> $astman2->sendcommand(Action => 'DBPut', Family => 'cnam', Key =>
> 'count', Val => '0');
> my @result102 = $astman2->sendcommand(Event => 'DBGetResponse');
> my $cnamreset2 = $result102[1];
>
> ##disconnect the manager connections##
> $astman1->disconnect;
> $astman2->disconnect;
>
> print "Total CNAM Count for last month is $cnamtotal\n\n";
> -end script
>
>
> The response from the 1.8.6.0 server is "Response Success Message
> Result will follow" but is seems the actual response is not pulled
> into $result2. The DBPut command works fine and I get a success
> response. I've searched through all the upgrade docs but nothing
> mentions command syntax changes.
>
The query syntax and result number changed from:
$astman2->sendcommand(Action => 'DBGet', Family => 'cnam', Key => 'count');
my @result2 = $astman2->sendcommand(Event => 'DBGetResponse');
my $cnamcount2 = "0$result2[7]";
To:
$astman2->command('database showkey cnam/count');
my @result2 = $astman2->sendcommand(Event => 'DBGetResponse');
my $cnamcount2 = "0$result2[5]";
JR
--
JR Richardson
Engineering for the Masses
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Re: [asterisk-users] making announcements
I was planing to use linux based softphone to have more control and flexibility From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder Sent: Thursday, October 06, 2011 3:42 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] making announcements On 10/05/2011 03:36 PM, Baha @ SH wrote: the phone hardware has to support it, and be setup to allow it (answer without ring and go to speakerphone mode) if it does its pretty straightforward to get working - but I have had issues in the past where when you hang up 100, the other phones still see each other as being offhook and connected so they never actually hang up again. Hello everyone I would like to know how possibly to make an announcement with asterisk. i.e. I have a phone registered with extension 100 when I dial (for example) 500, then all the phones with extension 501,502,.510 will automatically answers my call, and I speak my announcement. Any help in the issue very appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] making announcements
On 10/05/2011 03:36 PM, Baha @ SH wrote: the phone hardware has to support it, and be setup to allow it (answer without ring and go to speakerphone mode) if it does its pretty straightforward to get working - but I have had issues in the past where when you hang up 100, the other phones still see each other as being offhook and connected so they never actually hang up again. Hello everyone I would like to know how possibly to make an announcement with asterisk. i.e. I have a phone registered with extension 100 when I dial (for example) 500, then all the phones with extension 501,502,...510 will automatically answers my call, and I speak my announcement. Any help in the issue very appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] making announcements
look at page application Regards - Original Message - From: Baha @ SH To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, October 05, 2011 2:36 PM Subject: [asterisk-users] making announcements Hello everyone I would like to know how possibly to make an announcement with asterisk. i.e. I have a phone registered with extension 100 when I dial (for example) 500, then all the phones with extension 501,502,.510 will automatically answers my call, and I speak my announcement. Any help in the issue very appreciated. -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] making announcements
It depends on your Asterisk version and the type of phones you use. If both are acceptable, you can use SIPAddHeader to make the phones pick up and "become an intercom". Try some "google-fu" on this one. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Baha @ SH Sent: Wednesday, October 05, 2011 2:37 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] making announcements Hello everyone I would like to know how possibly to make an announcement with asterisk. i.e. I have a phone registered with extension 100 when I dial (for example) 500, then all the phones with extension 501,502,.510 will automatically answers my call, and I speak my announcement. Any help in the issue very appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] making announcements
Hello everyone I would like to know how possibly to make an announcement with asterisk. i.e. I have a phone registered with extension 100 when I dial (for example) 500, then all the phones with extension 501,502,.510 will automatically answers my call, and I speak my announcement. Any help in the issue very appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] parking lot
On 10/05/2011 02:03 PM, Danny Nicholas wrote: Depending on hardware and number of parking lots, could hints be used to let everyone know that a parking lot was just put into use by blind transfer? (I have a PERL web interface that does this kind of check for 1.4 but that probably wouldn't help OP). Yes, and I think I've seen patches to do that floating around somewhere. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Different revisions of Digium cards
Does anyone know if there is a resource to see what changes were made between different versions of Digium cards? For example, how different is a TE410P revision C when compared to a TE410P 5th generation card? I know there were changes made to the architecture to address IRQ issues, etc.. but when did those occur? In other words, is there a card revision changelog? -- Kyle Sexton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] parking lot
Depending on hardware and number of parking lots, could hints be used to let
everyone know that a parking lot was just put into use by blind transfer?
(I have a PERL web interface that does this kind of check for 1.4 but that
probably wouldn't help OP).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, October 05, 2011 1:59 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] parking lot
On 10/05/2011 11:31 AM, Keith Sloan wrote:
> Hey list.
>
> I am having some issues with a parking lot. I am looking for a way
> that the person can blind transfer to a parking lot. it works
> perfectly accept being an unattended transfer its the caller who hears
> the parking lot position. anyone have a work around to this.
>
> exten => 999,1,Set(CHANNEL(parkinglot)=Parkinglot-keith)
> exten => 999,n,Set(PARKINGLOT=Parkinglot_keith)
> exten => 999,n,Set(CALLCHAN=${CUT(BLINDTRANSFER,-,1)})
> exten => 999,n,ParkAndAnnounce(PARKED,60,${CALLCHAN})
>
>
> the ParkAndAnnounce seems to be failing. any suggestions?
Define 'failing'? Does the call not get parked?
If by 'failing' you mean that the parked channel is the one who hears the
parking lot number announcement, that's perfectly understandable when a
blind transfer is done to this extension. When this extension's code starts
executing after a blind transfer, the transferer's channel is *already
gone*; there is no way to announce the parking lot number to them.
The only solution here is to not announce the parking lot number at all if
the call was blind-transferred, but of course then nobody will know what
parking lot the call was parked in.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at
www.digium.com & www.asterisk.org
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Re: [asterisk-users] parking lot
On 10/05/2011 11:31 AM, Keith Sloan wrote:
Hey list.
I am having some issues with a parking lot. I am looking for a way that
the person can blind transfer to a parking lot. it works perfectly
accept being an unattended transfer its the caller who hears the parking
lot position. anyone have a work around to this.
exten => 999,1,Set(CHANNEL(parkinglot)=Parkinglot-keith)
exten => 999,n,Set(PARKINGLOT=Parkinglot_keith)
exten => 999,n,Set(CALLCHAN=${CUT(BLINDTRANSFER,-,1)})
exten => 999,n,ParkAndAnnounce(PARKED,60,${CALLCHAN})
the ParkAndAnnounce seems to be failing. any suggestions?
Define 'failing'? Does the call not get parked?
If by 'failing' you mean that the parked channel is the one who hears
the parking lot number announcement, that's perfectly understandable
when a blind transfer is done to this extension. When this extension's
code starts executing after a blind transfer, the transferer's channel
is *already gone*; there is no way to announce the parking lot number to
them.
The only solution here is to not announce the parking lot number at all
if the call was blind-transferred, but of course then nobody will know
what parking lot the call was parked in.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
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Re: [asterisk-users] Reduce the wav file size
On 10/05/2011 01:30 AM, mahesh katta wrote:
Hi list,
How to reduce the meetme wav file size in asterisk. how can I
automatically reduce this file size.
exten =>
_8600[1-2]XX,1(record),SetVar(MEETME_RECORDINGFILE=/var/spool/asterisk/meetme/conference_recording-${EPOCH}-${USER}-${TIMESTAMP}-${EXTEN});
exten => _8600[1-2]XX,2,Meetme,${EXTEN}|Fr
A WAV file is (generally) uncompressed audio; it's size cannot be
reduced except through compression, which would need to be reversed in
order to play back the file.
If you want to store recordings using less disk space, store in a
compressed format like GSM, G.729 or something else supported in Asterisk.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
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Re: [asterisk-users] Asterisk (Trixbox) - VirtualBox - Linux Host
someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. I just installed 3 Trixbox systems in KVM on Ubuntu. They're emergency PBX's for a few companies who lost their phone systems in a flood. They'll become real machines located on the customer premesis in the near future, but they've been running fine for a couple of weeks as virtual machines. One customer reported "gaps" in the hold music, but that was the only issue and I have no reason to suspect it's related to being virtual machine. I have not tried VirtualBox. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call pickup
hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? thanks -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme
hi, you are using pattern matching and not using the right syntax like that. exten => _520,1,answer like that. On 5 Oct 2011 21:47, "salaheddine elharit" wrote: > Hello list > > > > i have one question related to meetme,i have to providers with the first one > i put the number with 9 digit 520XX and all works without issue, with > the second i put just the last 3 numbers 500 with meetme there is nothing > > > > but when i put the last 3 numbers like below i can call my sip without any > problem, could you please inform me if the issue is related to my provider > of the issue come from asterisk > > > exten => 500,1,Dial(SIP/228, 30) > > extensions.conf > > first provider > exten => 520XX,1,Answer > exten => 520XX,n,Wait(4) > exten => 520XX,n,Meetme > = > second provider > > exten => 500,1,Answer > exten => 500,n,Wait(4) > exten => 500,n,Meetme > > there is no meetme with this one > > > > meetme.conf > > conf =>1234,5678 > > thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] parking lot
Sorry forgot to mention this is on an asterisk 1.6.2.13 installation
On 11-10-05 12:31 PM, Keith Sloan wrote:
Hey list.
I am having some issues with a parking lot. I am looking for a way
that the person can blind transfer to a parking lot. it works
perfectly accept being an unattended transfer its the caller who hears
the parking lot position. anyone have a work around to this.
exten => 999,1,Set(CHANNEL(parkinglot)=Parkinglot-keith)
exten => 999,n,Set(PARKINGLOT=Parkinglot_keith)
exten => 999,n,Set(CALLCHAN=${CUT(BLINDTRANSFER,-,1)})
exten => 999,n,ParkAndAnnounce(PARKED,60,${CALLCHAN})
the ParkAndAnnounce seems to be failing. any suggestions?
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[asterisk-users] parking lot
Hey list.
I am having some issues with a parking lot. I am looking for a way that
the person can blind transfer to a parking lot. it works perfectly
accept being an unattended transfer its the caller who hears the parking
lot position. anyone have a work around to this.
exten => 999,1,Set(CHANNEL(parkinglot)=Parkinglot-keith)
exten => 999,n,Set(PARKINGLOT=Parkinglot_keith)
exten => 999,n,Set(CALLCHAN=${CUT(BLINDTRANSFER,-,1)})
exten => 999,n,ParkAndAnnounce(PARKED,60,${CALLCHAN})
the ParkAndAnnounce seems to be failing. any suggestions?
--
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[asterisk-users] unsubscribe
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[asterisk-users] meetme
Hello list i have one question related to meetme,i have to providers with the first one i put the number with 9 digit 520XX and all works without issue, with the second i put just the last 3 numbers 500 with meetme there is nothing but when i put the last 3 numbers like below i can call my sip without any problem, could you please inform me if the issue is related to my provider of the issue come from asterisk exten => 500,1,Dial(SIP/228, 30) extensions.conf first provider exten => 520XX,1,Answer exten => 520XX,n,Wait(4) exten => 520XX,n,Meetme = second provider exten => 500,1,Answer exten => 500,n,Wait(4) exten => 500,n,Meetme there is no meetme with this one meetme.conf conf =>1234,5678 thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PRI hangup
> Sorry for the resend, but i don't have got any response, so i try to
> re-open the same problem.
>
> Hello all,
>
> Form 2-3 weeks i have some problems with incoming ISDN calls, it
> interrupts after 1-2 minutes of call. I have tried to debug this with
> pri set debug on span 1, i have noticied much of this messages:
>
> -- Timeout occured, restarting PRI
> q921.c:468 t200_expire: q921_state now is
> Q921_LINK_CONNECTION_RELEASED
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> Sending TEI management message 1, TEI=127
> Received MDL message
> TEI assiged to 71
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> Sending Set Asynchronous Balanced Mode Extended
> q921.c:211 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH
> -- Got UA from network peer Link up.
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> q921.c:805 q921_dchannel_up: q921_state now is
> Q921_LINK_CONNECTION_ESTABLISHED
> -- Timeout occured, restarting PRI
> q921.c:468 t200_expire: q921_state now is
> Q921_LINK_CONNECTION_RELEASED
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> Sending TEI management message 1, TEI=127
> Received MDL message
> TEI assiged to 72
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> Sending Set Asynchronous Balanced Mode Extended
> q921.c:211 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH
> -- Got UA from network peer Link up.
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> q921.c:805 q921_dchannel_up: q921_state now is
> Q921_LINK_CONNECTION_ESTABLISHED
>
> And there is a debug session of an hanged-up incoming call:
>
>
>
> > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
> (0) 0: 0 Location: Private network serving the local user (1)
> > Ext: 1 Progress Description: Inband
> information or appropriate pattern now available. (8) ]
> == Extension Changed 215[ext-local] new state Ringing for Notify User
> 202
> -- SIP/203-0017 is ringing
> -- SIP/206-0019 is ringing
> -- SIP/210-001a is ringing
> -- SIP/205-0018 is ringing
> -- SIP/201-0015 is ringing
> -- SIP/215-001b is ringing
> -- SIP/202-0016 is ringing
> -- SIP/201-0015 answered DAHDI/1-1
> == Extension Changed 201[ext-local] new state InUse for Notify User
> 202
> == Extension Changed 201[ext-local] new state InUse for Notify User
> 215
> == Extension Changed 215[ext-local] new state Idle for Notify User 202
> == Extension Changed 210[ext-local] new state Idle for Notify User 202
> == Extension Changed 210[ext-local] new state Idle for Notify User 215
> == Extension Changed 206[ext-local] new state Idle for Notify User 202
> == Extension Changed 206[ext-local] new state Idle for Notify User 215
> == Extension Changed 205[ext-local] new state Idle for Notify User 202
> == Extension Changed 205[ext-local] new state Idle for Notify User 215
> == Extension Changed 203[ext-local] new state Idle for Notify User 202
> == Extension Changed 203[ext-local] new state Idle for Notify User 215
> -- Executing [s@macro-auto-blkvm:1] Set("SIP/201-0015",
> "__MACRO_RESULT=") in new stack
> == Extension Changed 202[ext-local] new state Idle for Notify User 215
> -- Executing [s@macro-auto-blkvm:2] NoOp("SIP/201-0015",
> "Deleting: BLKVM/600/DAHDI/1-1 TRUE") in new stack
> -- Stopped music on hold on DAHDI/1-1
> q931.c:2951 q931_connect: call 93 on channel 1 enters state 8 (Connect
> Request)
> > Protocol Discriminator: Q.931 (8) len=11
> > Call Ref: len= 1 (reference 93/0x5D) (Terminator)
> > Message type: CONNECT (7)
> > [18 01 89]
> > Channel ID (len= 3) [ Ext: 1 IntID: Implicit Other Spare: 0
> Exclusive Dchan: 0
> >ChanSel: B1 channel
> ]
> > [1e 02 81 82]
> > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
> (0) 0: 0 Location: Private network serving the local user (1)
> > Ext: 1 Progress Description: Called
> equipment is non-ISDN. (2) ]
> < Protocol Discriminator: Q.931 (8) len=4
> < Call Ref: len= 1 (reference 93/0x5D) (Originator)
> < Message type: CONNECT ACKNOWLEDGE (15)
> q931.c:3711 q931_receive: call 93 on channel 1 enters state 10
> (Active)
> -- Got SABME from network peer.
> Sending Unnumbered Acknowledgement
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> q921.c:805 q921_dchannel_up: q921_state now is
> Q921_LINK_CONNECTION_ESTABLISHED
> < Protocol Discriminator: Q.931 (8) len=4
> < Call Ref: len= 1 (reference 93/0x5D) (Originator)
> < Message type: STATUS ENQUIRY (117)
> YYY Here we get reset YYY
> > Protocol Discriminator: Q.931 (8) len=7
> > Call Ref: len= 1 (reference 93/0x5D) (Terminator)
> > Message type: STATUS (125)
> > [14 01 00]
> > Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call
> state: Null (0)
> -- Got reject requesting packet 0... Retransmitting.
> < Protoc
Re: [asterisk-users] Asterisk PRI hangup
Sorry for the resend, but i don't have got any response, so i try to
re-open the same problem.
Hello all,
Form 2-3 weeks i have some problems with incoming ISDN calls, it
interrupts after 1-2 minutes of call. I have tried to debug this with
pri set debug on span 1, i have noticied much of this messages:
-- Timeout occured, restarting PRI
q921.c:468 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
Sending TEI management message 1, TEI=127
Received MDL message
TEI assiged to 71
q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
Sending Set Asynchronous Balanced Mode Extended
q921.c:211 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH
-- Got UA from network peer Link up.
q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:805 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
-- Timeout occured, restarting PRI
q921.c:468 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
Sending TEI management message 1, TEI=127
Received MDL message
TEI assiged to 72
q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
Sending Set Asynchronous Balanced Mode Extended
q921.c:211 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH
-- Got UA from network peer Link up.
q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:805 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
And there is a debug session of an hanged-up incoming call:
> Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: Private network serving the local user (1)
> Ext: 1 Progress Description: Inband
information or appropriate pattern now available. (8) ]
== Extension Changed 215[ext-local] new state Ringing for Notify User 202
-- SIP/203-0017 is ringing
-- SIP/206-0019 is ringing
-- SIP/210-001a is ringing
-- SIP/205-0018 is ringing
-- SIP/201-0015 is ringing
-- SIP/215-001b is ringing
-- SIP/202-0016 is ringing
-- SIP/201-0015 answered DAHDI/1-1
== Extension Changed 201[ext-local] new state InUse for Notify User 202
== Extension Changed 201[ext-local] new state InUse for Notify User 215
== Extension Changed 215[ext-local] new state Idle for Notify User 202
== Extension Changed 210[ext-local] new state Idle for Notify User 202
== Extension Changed 210[ext-local] new state Idle for Notify User 215
== Extension Changed 206[ext-local] new state Idle for Notify User 202
== Extension Changed 206[ext-local] new state Idle for Notify User 215
== Extension Changed 205[ext-local] new state Idle for Notify User 202
== Extension Changed 205[ext-local] new state Idle for Notify User 215
== Extension Changed 203[ext-local] new state Idle for Notify User 202
== Extension Changed 203[ext-local] new state Idle for Notify User 215
-- Executing [s@macro-auto-blkvm:1] Set("SIP/201-0015",
"__MACRO_RESULT=") in new stack
== Extension Changed 202[ext-local] new state Idle for Notify User 215
-- Executing [s@macro-auto-blkvm:2] NoOp("SIP/201-0015",
"Deleting: BLKVM/600/DAHDI/1-1 TRUE") in new stack
-- Stopped music on hold on DAHDI/1-1
q931.c:2951 q931_connect: call 93 on channel 1 enters state 8 (Connect
Request)
> Protocol Discriminator: Q.931 (8) len=11
> Call Ref: len= 1 (reference 93/0x5D) (Terminator)
> Message type: CONNECT (7)
> [18 01 89]
> Channel ID (len= 3) [ Ext: 1 IntID: Implicit Other Spare: 0
Exclusive Dchan: 0
>ChanSel: B1 channel
]
> [1e 02 81 82]
> Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: Private network serving the local user (1)
> Ext: 1 Progress Description: Called
equipment is non-ISDN. (2) ]
< Protocol Discriminator: Q.931 (8) len=4
< Call Ref: len= 1 (reference 93/0x5D) (Originator)
< Message type: CONNECT ACKNOWLEDGE (15)
q931.c:3711 q931_receive: call 93 on channel 1 enters state 10 (Active)
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement
q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:805 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
< Protocol Discriminator: Q.931 (8) len=4
< Call Ref: len= 1 (reference 93/0x5D) (Originator)
< Message type: STATUS ENQUIRY (117)
YYY Here we get reset YYY
> Protocol Discriminator: Q.931 (8) len=7
> Call Ref: len= 1 (reference 93/0x5D) (Terminator)
> Message type: STATUS (125)
> [14 01 00]
> Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call
state: Null (0)
-- Got reject requesting packet 0... Retransmitting.
< Protocol Discriminator: Q.931 (8) len=4
< Call Ref: len= 1 (reference 93/0x5D) (Originator)
< Message type: STAT
Re: [asterisk-users] Passive wait in dialplan?
So here's what I think about your scenario: CALL-FLOW 1- Call come in to asterisk (channel not answered) 2- Event is triggered and User decides what to do with call 3- On basis of what user decided a variable is set. 4- Asterisk on the base of that variable route the call further. If this is the intended behaviour I'd make the dialplan which would be something like. DIAL-PLAN ALGO 1- Progress() ; Won't Answer the channel and put the call in trying... mode. 2- Generate User Evnt 3- While(USERDECISION == "") 4- Endwhile 5- Execute anything on base of USERDECISION This has some limitation due to progress. GUI user needs to decide fast as progress will time-up and the caller will get NO_ANSWER from the system. Queue can be used to put call on wait until something is decided by GUI user but for that you'll have to use system resources and also answer the channel first. I hope some real expert here guide you in a better direction. On Wed, Oct 5, 2011 at 4:44 PM, Yaroslav Panych wrote: > Yes, something like that, but > hold"-state should not answer channel. answer command will be given > explicitly. or call can be transfered to Dial command, etc. > > 2011/10/5 Sammy Govind : > > Can you please explain what you are trying to do? What I've perceived > from > > this thread is that you want to put call on hold (passively as in no > > resources usage) and then on the base of some User's input from UI > proceed > > with the call accordingly !! > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reinvite dialplan application [Was: OT - SIP - Toggle to autoanswer after ringing]
2011/10/5 Steve Davies > On 5 October 2011 10:21, Nasir Iqbal wrote: > > You can do this by an AMI based transfer (Redirect) to Local channel, and > > then in local channel's dialplan you need to add your desired custom sip > > header followed by a dial command. > > Nasir Iqbal > > > > ICT Innovations > > http://www.ictinnovations.com/ > > > > Broadcom invented some SIP NOTIFY extensions to cover this case - > Several "open" SIP handsets support the Broadcom extensions, which > revolve around sending a NOTIFY "Event: hold" or "Event: talk" > > Asterisk does not support these NOTIFY messages at present, though I > expect they could be added reasonably simply. > Googling with "NOTIFY "Event: hold", I found a doc (Broadworks sip access side extension) which exactly match what I'm after : a couple of events to "start or resume autoanswer". Thank you very much for sharing this ! > > Regards, > Steve > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passive wait in dialplan?
Yes, something like that, but hold"-state should not answer channel. answer command will be given explicitly. or call can be transfered to Dial command, etc. 2011/10/5 Sammy Govind : > Can you please explain what you are trying to do? What I've perceived from > this thread is that you want to put call on hold (passively as in no > resources usage) and then on the base of some User's input from UI proceed > with the call accordingly !! > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reinvite dialplan application [Was: OT - SIP - Toggle to autoanswer after ringing]
2011/10/5 Nasir Iqbal > You can do this by an AMI based transfer (Redirect) to Local channel, and > then in local channel's dialplan you need to add your desired custom sip > header followed by a dial command. > > Nasir Iqbal > > ICT Innovations > http://www.ictinnovations.com/ > > Thanks for the tip : I'll give it a try ASAP (and report here)! Thanks again ! > > > On Wed, Oct 5, 2011 at 11:36 AM, Olivier wrote: > >> >> 2011/10/4 Olivier >> >>> Hi, >>> >>> Has anyone heard (or read) about an existing or emerging standard >>> targeting the following feature : >>> 1. a SIP handset receives an incoming call >>> 2. this handset starts ringing >>> 3. then it receives an update asking to autoanswer the ringing call. >>> >>> This feature would help to build software panels complementing or >>> replicating hard phones GUI. >>> >>> (I know you can work around such feature using conference rooms or >>> dealing with hard phones API (really ?) but in order to keep Queue log >>> accurate, this feature would be useful). >>> >>> Cheers >>> >> >> Hi, >> >> In my quest to allow a software panel to ask an hardphone to answer an >> incoming call without touching the hardphone itself, I'm wondering if a >> Reinvite application could exist. >> >> I'm thinking about the following use case : >> >> Alice is calling Bob >> Bob's phone starts to ring >> Bob's GUI app also shows the incoming call asking him if he prefers to >> reject, transfer or answer the call >> With the GUI app, Bob replies he whishes to reply >> Asterisk reinvites Bob's phone with autoanswer option >> Bob's phone answers Alice phone >> >> Thoughts ? >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold
thanks for your replay i give the permissions 777 to this file moh1 and i
still have the same issue
best regards
2011/10/5 Sammy Govind
> Give that moh1 directory permissions, I once had similar issue that same
> files being placed in default moh directory were played but making a new
> call and directory couldn't play anything. So I fixed that by granting
> directory permissions.
>
>
> On Wed, Oct 5, 2011 at 2:25 PM, salaheddine elharit <
> salah.elharit...@gmail.com> wrote:
>
>> Hi
>>
>> yes i have noticed the same result when i play a file like the default i
>> can hear the music but when i play another file there is no sound
>>
>> about your question danny :yes i have created a file in
>> /var/lib/asterisk/moh1
>>
>> and i configure in musiconhold.conf like below
>>
>> [default1]
>> mode=files
>> directory=/var/lib/asterisk/moh1
>>
>> 2011/10/4 Kevin Oravits
>>
>>>I’ve noticed on our system the sound files have to be in an exact
>>> format for Asterisk to play them.
>>>
>>> Bit Rate: 128kbps
>>>
>>> Audio sample size: 16 bit
>>>
>>> Channels: 1(mono)
>>>
>>> Audio Sample rate: 8kHz
>>>
>>> Audio format: PCM
>>>
>>> ** **
>>>
>>> I actually downloaded a program and remixed the audio files to match
>>> these settings. Before that, I couldn’t get my Asterisk to play any
>>> non-standard music.
>>>
>>> ** **
>>>
>>> *Kevin Oravits *
>>>
>>> ** **
>>>
>>> *From:* Danny Nicholas [mailto:da...@debsinc.com]
>>> *Sent:* Tuesday, October 04, 2011 11:48 AM
>>>
>>> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
>>> *Subject:* Re: [asterisk-users] music on hold
>>>
>>> ** **
>>>
>>> You have files in /var/lib/asterisk/moh1?
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
>>> elharit
>>> *Sent:* Tuesday, October 04, 2011 12:49 PM
>>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>>> *Subject:* [asterisk-users] music on hold
>>>
>>> ** **
>>>
>>> i configure new music on hold like below in order to play music for
>>> outbond calls
>>>
>>> i want tp play a music until answer form customer
>>>
>>> [default1]
>>> mode=files
>>> directory=/var/lib/asterisk/moh1
>>>
>>> exten => 0678XX,1,Set(CALLERID(number)=520XX)
>>> exten =>
>>> 0678XX,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
>>> exten => 0678XX,n,Dial(Zap/g1/${EXTEN},,m(default1))
>>> exten => 0678XX,n,Hangup()
>>>
>>>
>>> when i put the default music i can listen without issue but when i put
>>> another music .wav Or gsm or Mp3
>>>
>>> there is no music there is just the ringing
>>>
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>> http://www.asterisk.org/hello
>>>
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>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _
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>
>
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Re: [asterisk-users] Passive wait in dialplan?
Can you please explain what you are trying to do? What I've perceived from
this thread is that you want to put call on hold (passively as in no
resources usage) and then on the base of some User's input from UI proceed
with the call accordingly !!
On Wed, Oct 5, 2011 at 3:33 PM, Yaroslav Panych wrote:
> I don't know much about queues, but if channel enter into queue it
> should not change its state. I.e. not answer, no moh, no interacting
> with user input(DTMF). Less I use unknown helpers, better my
> configuration is.
> Second issue which can appear using queues - its async state. User can
> issue 2 serial commands, and I should have synchronisation tools. In
> dialplan I using UserEvent application - which issues event in AMI,
> with given data headers. Queue - is there any possibility to customise
> queue join(or like) AMI event? Without patches(I already have made
> some patches to core and wrote additional module to make * work as I
> require).
>
>
> 2011/10/5 Nasir Iqbal :
> > What about waiting in "queues"?
> > Nasir Iqbal
> >
> > ICT Innovations
> > http://www.ictinnovations.com/
> >
> >
> >
> > On Wed, Oct 5, 2011 at 1:35 PM, Yaroslav Panych
> wrote:
> >>
> >> Hello, everyone
> >>
> >> Here part of my dialplan context
> >> [globals]
> >> CMD_NOOP=0
> >> CMD_DOSTUFF1=1
> >> CMD_DOSTUFF2=2
> >> CMD_DOSTUFF3=2
> >>
> >> [blah-context]
> >> same => n,Set(COMMAND=${CMD_NOOP})
> >> same => n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
> >> same =>
> >>
> n(COMMAND_SWITCH),GoToIf($["${COMMAND}"="${CMD_DOSTUFF1}"]?LBL_DO_STUFF1)
> >> same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF2}"]?LBL_DO_STUFF2)
> >> same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF3}"]?LBL_DO_STUFF3)
> >> same => n,Wait(0.2)
> >> same => n,GoTo(COMMAND_SWITCH)
> >> same => n,NoOp(--- NOT REACHED ---)
> >>
> >> UserEvent sends blah-event via AMI to high-level UI, user makes
> >> decision and issues some command via Action:SetVar, then dialplan
> >> continues to work.
> >>
> >> The problem is, in dialplan there is an active wait loop, i.e. waiting
> >> mechanism which rapidly checks some var(consuming processor resources
> >> and flooding logs). Is it possible to create passive waiting loop
> >> within current abilities of Asterisk 1.8?
> >>
> >> regards, Yaroslav
> >>
> >> --
> >> _
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >> http://www.asterisk.org/hello
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > --
> > _
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> > http://www.asterisk.org/hello
> >
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> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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Re: [asterisk-users] Passive wait in dialplan?
I don't know much about queues, but if channel enter into queue it
should not change its state. I.e. not answer, no moh, no interacting
with user input(DTMF). Less I use unknown helpers, better my
configuration is.
Second issue which can appear using queues - its async state. User can
issue 2 serial commands, and I should have synchronisation tools. In
dialplan I using UserEvent application - which issues event in AMI,
with given data headers. Queue - is there any possibility to customise
queue join(or like) AMI event? Without patches(I already have made
some patches to core and wrote additional module to make * work as I
require).
2011/10/5 Nasir Iqbal :
> What about waiting in "queues"?
> Nasir Iqbal
>
> ICT Innovations
> http://www.ictinnovations.com/
>
>
>
> On Wed, Oct 5, 2011 at 1:35 PM, Yaroslav Panych wrote:
>>
>> Hello, everyone
>>
>> Here part of my dialplan context
>> [globals]
>> CMD_NOOP=0
>> CMD_DOSTUFF1=1
>> CMD_DOSTUFF2=2
>> CMD_DOSTUFF3=2
>>
>> [blah-context]
>> same => n,Set(COMMAND=${CMD_NOOP})
>> same => n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
>> same =>
>> n(COMMAND_SWITCH),GoToIf($["${COMMAND}"="${CMD_DOSTUFF1}"]?LBL_DO_STUFF1)
>> same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF2}"]?LBL_DO_STUFF2)
>> same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF3}"]?LBL_DO_STUFF3)
>> same => n,Wait(0.2)
>> same => n,GoTo(COMMAND_SWITCH)
>> same => n,NoOp(--- NOT REACHED ---)
>>
>> UserEvent sends blah-event via AMI to high-level UI, user makes
>> decision and issues some command via Action:SetVar, then dialplan
>> continues to work.
>>
>> The problem is, in dialplan there is an active wait loop, i.e. waiting
>> mechanism which rapidly checks some var(consuming processor resources
>> and flooding logs). Is it possible to create passive waiting loop
>> within current abilities of Asterisk 1.8?
>>
>> regards, Yaroslav
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _
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> http://www.asterisk.org/hello
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Passive wait in dialplan?
What about waiting in "queues"?
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Wed, Oct 5, 2011 at 1:35 PM, Yaroslav Panych wrote:
> Hello, everyone
>
> Here part of my dialplan context
> [globals]
> CMD_NOOP=0
> CMD_DOSTUFF1=1
> CMD_DOSTUFF2=2
> CMD_DOSTUFF3=2
>
> [blah-context]
> same => n,Set(COMMAND=${CMD_NOOP})
> same => n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
> same =>
> n(COMMAND_SWITCH),GoToIf($["${COMMAND}"="${CMD_DOSTUFF1}"]?LBL_DO_STUFF1)
> same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF2}"]?LBL_DO_STUFF2)
> same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF3}"]?LBL_DO_STUFF3)
> same => n,Wait(0.2)
> same => n,GoTo(COMMAND_SWITCH)
> same => n,NoOp(--- NOT REACHED ---)
>
> UserEvent sends blah-event via AMI to high-level UI, user makes
> decision and issues some command via Action:SetVar, then dialplan
> continues to work.
>
> The problem is, in dialplan there is an active wait loop, i.e. waiting
> mechanism which rapidly checks some var(consuming processor resources
> and flooding logs). Is it possible to create passive waiting loop
> within current abilities of Asterisk 1.8?
>
> regards, Yaroslav
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Reinvite dialplan application [Was: OT - SIP - Toggle to autoanswer after ringing]
On 5 October 2011 10:21, Nasir Iqbal wrote: > You can do this by an AMI based transfer (Redirect) to Local channel, and > then in local channel's dialplan you need to add your desired custom sip > header followed by a dial command. > Nasir Iqbal > > ICT Innovations > http://www.ictinnovations.com/ > Broadcom invented some SIP NOTIFY extensions to cover this case - Several "open" SIP handsets support the Broadcom extensions, which revolve around sending a NOTIFY "Event: hold" or "Event: talk" Asterisk does not support these NOTIFY messages at present, though I expect they could be added reasonably simply. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold
Give that moh1 directory permissions, I once had similar issue that same
files being placed in default moh directory were played but making a new
call and directory couldn't play anything. So I fixed that by granting
directory permissions.
On Wed, Oct 5, 2011 at 2:25 PM, salaheddine elharit <
salah.elharit...@gmail.com> wrote:
> Hi
>
> yes i have noticed the same result when i play a file like the default i
> can hear the music but when i play another file there is no sound
>
> about your question danny :yes i have created a file in
> /var/lib/asterisk/moh1
>
> and i configure in musiconhold.conf like below
>
> [default1]
> mode=files
> directory=/var/lib/asterisk/moh1
>
> 2011/10/4 Kevin Oravits
>
>> I’ve noticed on our system the sound files have to be in an exact format
>> for Asterisk to play them.
>>
>> Bit Rate: 128kbps
>>
>> Audio sample size: 16 bit
>>
>> Channels: 1(mono)
>>
>> Audio Sample rate: 8kHz
>>
>> Audio format: PCM
>>
>> ** **
>>
>> I actually downloaded a program and remixed the audio files to match these
>> settings. Before that, I couldn’t get my Asterisk to play any non-standard
>> music.
>>
>> ** **
>>
>> *Kevin Oravits *
>>
>> ** **
>>
>> *From:* Danny Nicholas [mailto:da...@debsinc.com]
>> *Sent:* Tuesday, October 04, 2011 11:48 AM
>>
>> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
>> *Subject:* Re: [asterisk-users] music on hold
>>
>> ** **
>>
>> You have files in /var/lib/asterisk/moh1?
>>
>> ** **
>>
>> ** **
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
>> elharit
>> *Sent:* Tuesday, October 04, 2011 12:49 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] music on hold
>>
>> ** **
>>
>> i configure new music on hold like below in order to play music for
>> outbond calls
>>
>> i want tp play a music until answer form customer
>>
>> [default1]
>> mode=files
>> directory=/var/lib/asterisk/moh1
>>
>> exten => 0678XX,1,Set(CALLERID(number)=520XX)
>> exten =>
>> 0678XX,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
>> exten => 0678XX,n,Dial(Zap/g1/${EXTEN},,m(default1))
>> exten => 0678XX,n,Hangup()
>>
>>
>> when i put the default music i can listen without issue but when i put
>> another music .wav Or gsm or Mp3
>>
>> there is no music there is just the ringing
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> http://www.asterisk.org/hello
>
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>
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Re: [asterisk-users] music on hold
Hi
yes i have noticed the same result when i play a file like the default i can
hear the music but when i play another file there is no sound
about your question danny :yes i have created a file in
/var/lib/asterisk/moh1
and i configure in musiconhold.conf like below
[default1]
mode=files
directory=/var/lib/asterisk/moh1
2011/10/4 Kevin Oravits
> I’ve noticed on our system the sound files have to be in an exact format
> for Asterisk to play them.
>
> Bit Rate: 128kbps
>
> Audio sample size: 16 bit
>
> Channels: 1(mono)
>
> Audio Sample rate: 8kHz
>
> Audio format: PCM
>
> ** **
>
> I actually downloaded a program and remixed the audio files to match these
> settings. Before that, I couldn’t get my Asterisk to play any non-standard
> music.
>
> ** **
>
> *Kevin Oravits *
>
> ** **
>
> *From:* Danny Nicholas [mailto:da...@debsinc.com]
> *Sent:* Tuesday, October 04, 2011 11:48 AM
>
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* Re: [asterisk-users] music on hold
>
> ** **
>
> You have files in /var/lib/asterisk/moh1?
>
> ** **
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Tuesday, October 04, 2011 12:49 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] music on hold
>
> ** **
>
> i configure new music on hold like below in order to play music for outbond
> calls
>
> i want tp play a music until answer form customer
>
> [default1]
> mode=files
> directory=/var/lib/asterisk/moh1
>
> exten => 0678XX,1,Set(CALLERID(number)=520XX)
> exten => 0678XX,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
> exten => 0678XX,n,Dial(Zap/g1/${EXTEN},,m(default1))
> exten => 0678XX,n,Hangup()
>
>
> when i put the default music i can listen without issue but when i put
> another music .wav Or gsm or Mp3
>
> there is no music there is just the ringing
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Reinvite dialplan application [Was: OT - SIP - Toggle to autoanswer after ringing]
You can do this by an AMI based transfer (Redirect) to Local channel, and then in local channel's dialplan you need to add your desired custom sip header followed by a dial command. Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Wed, Oct 5, 2011 at 11:36 AM, Olivier wrote: > > 2011/10/4 Olivier > >> Hi, >> >> Has anyone heard (or read) about an existing or emerging standard >> targeting the following feature : >> 1. a SIP handset receives an incoming call >> 2. this handset starts ringing >> 3. then it receives an update asking to autoanswer the ringing call. >> >> This feature would help to build software panels complementing or >> replicating hard phones GUI. >> >> (I know you can work around such feature using conference rooms or dealing >> with hard phones API (really ?) but in order to keep Queue log accurate, >> this feature would be useful). >> >> Cheers >> > > Hi, > > In my quest to allow a software panel to ask an hardphone to answer an > incoming call without touching the hardphone itself, I'm wondering if a > Reinvite application could exist. > > I'm thinking about the following use case : > > Alice is calling Bob > Bob's phone starts to ring > Bob's GUI app also shows the incoming call asking him if he prefers to > reject, transfer or answer the call > With the GUI app, Bob replies he whishes to reply > Asterisk reinvites Bob's phone with autoanswer option > Bob's phone answers Alice phone > > Thoughts ? > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Passive wait in dialplan
Hello, everyone
Here part of my dialplan context
[globals]
CMD_NOOP=0
CMD_DOSTUFF1=1
CMD_DOSTUFF2=2
CMD_DOSTUFF3=2
[blah-context]
same => n,Set(COMMAND=${CMD_NOOP})
same => n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
same =>
n(COMMAND_SWITCH),GoToIf($["${COMMAND}"="${CMD_DOSTUFF1}"]?LBL_DO_STUFF1)
same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF2}"]?LBL_DO_STUFF2)
same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF3}"]?LBL_DO_STUFF3)
same => n,Wait(0.2)
same => n,GoTo(
regards, Yaroslav
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[asterisk-users] Passive wait in dialplan?
Hello, everyone
Here part of my dialplan context
[globals]
CMD_NOOP=0
CMD_DOSTUFF1=1
CMD_DOSTUFF2=2
CMD_DOSTUFF3=2
[blah-context]
same => n,Set(COMMAND=${CMD_NOOP})
same => n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
same =>
n(COMMAND_SWITCH),GoToIf($["${COMMAND}"="${CMD_DOSTUFF1}"]?LBL_DO_STUFF1)
same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF2}"]?LBL_DO_STUFF2)
same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF3}"]?LBL_DO_STUFF3)
same => n,Wait(0.2)
same => n,GoTo(COMMAND_SWITCH)
same => n,NoOp(--- NOT REACHED ---)
UserEvent sends blah-event via AMI to high-level UI, user makes
decision and issues some command via Action:SetVar, then dialplan
continues to work.
The problem is, in dialplan there is an active wait loop, i.e. waiting
mechanism which rapidly checks some var(consuming processor resources
and flooding logs). Is it possible to create passive waiting loop
within current abilities of Asterisk 1.8?
regards, Yaroslav
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Re: [asterisk-users] Beep file with Record
Yes, That was the solution. Thanks. -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Jeroen Eeuwes Verzonden: 05-10-2011 10:15 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Beep file with Record Hi Arjan, > I try to change de astdatadir into /var/lib/asterisk/ > But when I restart asterisk and I look at the settings in the CLI I still see > Data directory: /usr/share/asterisk At least that explains why it can't find your beep-file. It is looking in /usr/share/asterisk and not /var/lib/asterisk. If your asterisk.conf says this: [directories](!) ; remove the (!) to enable this you should remove the (!) to enable the alternate directories in asterisk.conf so it should only say this: [directories] Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beep file with Record
Hi Arjan, > I try to change de astdatadir into /var/lib/asterisk/ > But when I restart asterisk and I look at the settings in the CLI I still see > Data directory: /usr/share/asterisk At least that explains why it can't find your beep-file. It is looking in /usr/share/asterisk and not /var/lib/asterisk. If your asterisk.conf says this: [directories](!) ; remove the (!) to enable this you should remove the (!) to enable the alternate directories in asterisk.conf so it should only say this: [directories] Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beep file with Record
These are the directories which I gave in asterisk.conf astetcdir => /etc/asterisk astmoddir => /usr/lib64/asterisk/modules astvarlibdir => /usr/share/asterisk astdbdir => /var/spool/asterisk astkeydir => /var/lib/asterisk astdatadir => /usr/share/asterisk astagidir => /usr/share/asterisk/agi-bin astspooldir => /var/spool/asterisk astrundir => /var/run/asterisk astlogdir => /var/log/asterisk I try to change de astdatadir into /var/lib/asterisk/ But when I restart asterisk and I look at the settings in the CLI I still see Data directory: /usr/share/asterisk How can I reload the new settings? -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Jeroen Eeuwes Verzonden: 05-10-2011 09:50 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Beep file with Record Hi Arjan, > I also try to place the voicefile in the directory /var/lib/asterisk/sounds/ > and /var/lib/asterisk/sounds/applications/ of but without any success. Just for double-checking, but what directory is listed as the astdatadir in asterisk.conf? Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beep file with Record
On 10/5/11 9:50 AM, Jeroen Eeuwes wrote: > Hi Arjan, > >> I also try to place the voicefile in the directory /var/lib/asterisk/sounds/ >> and /var/lib/asterisk/sounds/applications/ of but without any success. > > Just for double-checking, but what directory is listed as the > astdatadir in asterisk.conf? And if that still doesn't give a clue where Asterisk is looking for beep, monitor the asterisk executable with strace and grep its output for beep and that should point to your problem. You might need a little while to figure strace out but that is the way to be absolutely sure what Asterisk is trying to do. Everything else is just guessing. -- Andreas Sikkema -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beep file with Record
The alaw extension is bugging me.. can you locate the default beep.gsm /beep.wav file in asterisk sounds directory !? Also check the output of *core show file formats* *core show translation* Also find out the codec of the established call.! On Wed, Oct 5, 2011 at 12:50 PM, Jeroen Eeuwes wrote: > Hi Arjan, > > > I also try to place the voicefile in the directory > /var/lib/asterisk/sounds/ > > and /var/lib/asterisk/sounds/applications/ of but without any success. > > Just for double-checking, but what directory is listed as the > astdatadir in asterisk.conf? > > Best regards, > Jeroen Eeuwes > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk (Trixbox) - VirtualBox - Linux Host
> someone have been installed Asterisk (Trixbox) on VirtualBox which is > installed on a Linux host (Ubuntu server 10.04 specifically). > > > I want to know if it is convenient or not, and the reaseons if i should on > shouldn't do it. > > > Thanks in advance.! > > > > -- > Esteban L. Cacavelos de Amoriza > Cel: 0981 220 429 > > -- > I installed and used Elastix 2.0.3 on VirtualBox 4.x (64bit) but I were constantly troubled by high CPU usage and performance issues. I am not a virtualization / Asterisk expert so I may have missed some aspect of settings or configurations. My general reading on various forums seemed to indicate that VirtualBox is still not the best platform real time application like asterisk. My 2 cents -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beep file with Record
Hi Arjan, > I also try to place the voicefile in the directory /var/lib/asterisk/sounds/ > and /var/lib/asterisk/sounds/applications/ of but without any success. Just for double-checking, but what directory is listed as the astdatadir in asterisk.conf? Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beep file with Record
Oke, I tried this, but sorry
-- Executing [s@servicelijn:91] Set("CAPI/ISDN1#02/318647615-3e",
"CHANNEL(language)=en") in new stack
-- Executing [s@servicelijn:92] Set("CAPI/ISDN1#02/318647615-3e",
"A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/1317800460.74")
in new stack
-- Executing [s@servicelijn:93] Record("CAPI/ISDN1#02/318647615-3e",
"/var/lib/asterisk/sounds/recordings/serviceline/1317800460.74.wav,0,60") in
new stack
[Oct 5 09:41:03] WARNING[18963]: file.c:644 ast_openstream_full: File beep
does not exist in any format
[Oct 5 09:41:03] WARNING[18963]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory
[Oct 5 09:41:03] WARNING[18963]: app_record.c:281 record_exec: ast_streamfile
failed on CAPI/ISDN1#02/318647615-3e
== Spawn extension (servicelijn, s, 93) exited non-zero on
'CAPI/ISDN1#02/318647615-3e'
This is my conf.file
exten => s,n,Set(CHANNEL(language)=en)
exten =>
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID})
exten => s,n,Record(${A_serviceline_file}.wav,0,60)
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Sammy Govind
Verzonden: 05-10-2011 09:32
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
hmmm...what i'm saying is this
exten => s,n,Set(CHANNEL(language)=en))
exten =>
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten => s,n,Record(${A_serviceline_file}.wav,0,60)
exten => s,n,Set(CHANNEL(language)=nl))
On Wed, Oct 5, 2011 at 12:29 PM, Arjan Kroon | Mobillion
mailto:arjan.kr...@mobillion.nl>> wrote:
Yes I already try this (only with language nl)
exten => s,n,Set(CHANNEL(language)=nl))
I also try to place the voicefile in the directory /var/lib/asterisk/sounds/
and /var/lib/asterisk/sounds/applications/ of but without any success.
Van:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Namens Sammy Govind
Verzonden: 05-10-2011 09:26
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
Since you've changed the language (sound directory) So just as a test change
the language back to "en" and if it goes well revert back language after the
recording.
On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion
mailto:arjan.kr...@mobillion.nl>> wrote:
CLI::
-- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
"/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60") in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
does not exist in any format
[Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory
[Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec: ast_streamfile
failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =>
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten => s,n,Record(${A_serviceline_file}.wav,0,60)
I don't call the beep file in my dialplan.
Van:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Namens Sammy Govind
Verzonden: 05-10-2011 09:04
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
How are you calling the beep.alaw from the dialplan?
paste the relevant dialplan here and corresponding CLI logs.
On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
mailto:arjan.kr...@mobillion.nl>> wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten => s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Namens Danny Nicholas
Verzonden: 04-10-2011 17:16
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two "problems" here. Problem 1 is that you are using the alaw codec, so
it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in my
head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this is
just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2 characte
Re: [asterisk-users] Beep file with Record
Sorry:
*exten => s,n,Set(CHANNEL(language)=en)*
and
exten => s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/
recordings/serviceline/${UNIQUEID*}*)
NOT
*exten => s,n,Set(CHANNEL(language)=en))*
exten =>
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
On Wed, Oct 5, 2011 at 12:31 PM, Sammy Govind wrote:
> hmmm...what i'm saying is this
>
> *exten => s,n,Set(CHANNEL(language)=en))*
>
> exten =>
> s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
>
>
> exten => s,n,Record(${A_serviceline_file}.wav,0,60)
> *exten => s,n,Set(CHANNEL(language)=nl))*
> *
> *
>
>
> On Wed, Oct 5, 2011 at 12:29 PM, Arjan Kroon | Mobillion <
> arjan.kr...@mobillion.nl> wrote:
>
>> Yes I already try this (only with language nl)
>>
>> exten => s,n,Set(CHANNEL(language)=nl))
>>
>> ** **
>>
>> I also try to place the voicefile in the directory
>> /var/lib/asterisk/sounds/ and /var/lib/asterisk/sounds/applications/ of but
>> without any success.
>>
>> ** **
>>
>> *Van:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *Namens *Sammy Govind
>> *Verzonden:* 05-10-2011 09:26
>>
>> *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Onderwerp:* Re: [asterisk-users] Beep file with Record
>>
>> ** **
>>
>> Since you've changed the language (sound directory) So just as a test
>> change the language back to "en" and if it goes well revert back language
>> after the recording.
>>
>> ** **
>>
>> On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion <
>> arjan.kr...@mobillion.nl> wrote:
>>
>> CLI::
>>
>> -- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
>> "/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60") in
>> new stack
>>
>> [Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File
>> beep does not exist in any format
>>
>> [Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to
>> open beep (format 0x8 (alaw)): No such file or directory
>>
>> [Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec:
>> ast_streamfile failed on CAPI/ISDN1#02/318647615-37
>>
>>
>>
>> In de Conf file I use the following command:
>>
>> exten =>
>> s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
>>
>>
>> exten => s,n,Record(${A_serviceline_file}.wav,0,60)
>>
>> I don’t call the beep file in my dialplan.
>>
>>
>>
>>
>>
>> *Van:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *Namens *Sammy Govind
>> *Verzonden:* 05-10-2011 09:04
>>
>>
>> *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Onderwerp:* Re: [asterisk-users] Beep file with Record
>>
>>
>>
>> How are you calling the beep.alaw from the dialplan?
>>
>> paste the relevant dialplan here and corresponding CLI logs.
>>
>>
>>
>> On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion <
>> arjan.kr...@mobillion.nl> wrote:
>>
>> I placed a beep.alaw file in de directory, but I get the same result.
>>
>> Also I try to set the language just with two characters.
>> (exten => s,n,Set(CHANNEL(language)=nl))
>> And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
>> beep.alaw.
>> But with this also I get also the same result.
>>
>>
>> -Oorspronkelijk bericht-
>> Van: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
>>
>> Verzonden: 04-10-2011 17:16
>>
>> Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>> Onderwerp: Re: [asterisk-users] Beep file with Record
>>
>> I see two "problems" here. Problem 1 is that you are using the alaw
>> codec, so it seems to me that you need this file to exist -
>> /var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in
>> my head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this
>> is just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2
>> characters, not xx/ (nl/fvdb) (feel free to correct my assumption that
>> language has not been expanded beyond the 2 character limitation)?
>>
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
>> Mobillion
>> Sent: Tuesday, October 04, 2011 9:43 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Beep file with Record
>>
>> Yes,
>>
>> In the code I use set the language
>> exten => s,n,Set(CHANNEL(language)=nl/fvdb)
>>
>> So therefore I try also to place the file in the directory
>> /var/lib/asterisk/sounds/nl/fvdb/
>>
>>
>> -Oorspronkelijk bericht-
>> Van: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] Namens Andrew Latham
>> Verzonden:
Re: [asterisk-users] Beep file with Record
can you post the while dialplan? it seems cropped somewhere as i dont' see it
starting or ending anywhere.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 5 Oct 2011 12:31:49 +0500
From: govoi...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Beep file with Record
hmmm...what i'm saying is this
exten => s,n,Set(CHANNEL(language)=en))
exten =>
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)exten
=> s,n,Record(${A_serviceline_file}.wav,0,60)
exten => s,n,Set(CHANNEL(language)=nl))
On Wed, Oct 5, 2011 at 12:29 PM, Arjan Kroon | Mobillion
wrote:
Yes I already try this (only with language nl)
exten => s,n,Set(CHANNEL(language)=nl))
I also try to place the voicefile in the directory /var/lib/asterisk/sounds/
and /var/lib/asterisk/sounds/applications/ of but without any success.
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Sammy Govind
Verzonden: 05-10-2011 09:26
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
Since you've changed the language (sound directory) So just as a test change
the language back to "en" and if it goes well revert back language after the
recording.
On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion
wrote:
CLI::-- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
"/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60") in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
does not exist in any format [Oct 4 16:19:38] WARNING[13370]: file.c:950
ast_streamfile: Unable to open beep (format 0x8 (alaw)): No such file or
directory
[Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec: ast_streamfile
failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:exten =>
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten => s,n,Record(${A_serviceline_file}.wav,0,60)I don’t call the beep file
in my dialplan.
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Sammy Govind
Verzonden: 05-10-2011 09:04
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
How are you calling the beep.alaw from the dialplan?paste the relevant
dialplan here and corresponding CLI logs.
On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten => s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 17:16Aan: 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two "problems" here. Problem 1 is that you are using the alaw codec, so
it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in my
head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this is
just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2 characters,
not xx/ (nl/fvdb) (feel free to correct my assumption that language has not
been expanded beyond the 2 character limitation)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
Mobillion
Sent: Tuesday, October 04, 2011 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Beep file with Record
Yes,
In the code I use set the language
exten => s,n,Set(CHANNEL(language)=nl/fvdb)
So therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Andrew Latham
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
wrote:
> This is my complete CLI logging
>
> -- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
> "/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
> 0") in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
> ast_openstream_full: File beep does not exist in any format [Oct 4
> 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
> be
Re: [asterisk-users] Beep file with Record
hmmm...what i'm saying is this
*exten => s,n,Set(CHANNEL(language)=en))*
exten =>
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten => s,n,Record(${A_serviceline_file}.wav,0,60)
*exten => s,n,Set(CHANNEL(language)=nl))*
*
*
On Wed, Oct 5, 2011 at 12:29 PM, Arjan Kroon | Mobillion <
arjan.kr...@mobillion.nl> wrote:
> Yes I already try this (only with language nl)
>
> exten => s,n,Set(CHANNEL(language)=nl))
>
> ** **
>
> I also try to place the voicefile in the directory
> /var/lib/asterisk/sounds/ and /var/lib/asterisk/sounds/applications/ of but
> without any success.
>
> ** **
>
> *Van:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *Namens *Sammy Govind
> *Verzonden:* 05-10-2011 09:26
>
> *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Onderwerp:* Re: [asterisk-users] Beep file with Record
>
> ** **
>
> Since you've changed the language (sound directory) So just as a test
> change the language back to "en" and if it goes well revert back language
> after the recording.
>
> ** **
>
> On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion <
> arjan.kr...@mobillion.nl> wrote:
>
> CLI::
>
> -- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
> "/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60") in
> new stack
>
> [Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
> does not exist in any format
>
> [Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
> beep (format 0x8 (alaw)): No such file or directory
>
> [Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec:
> ast_streamfile failed on CAPI/ISDN1#02/318647615-37
>
>
>
> In de Conf file I use the following command:
>
> exten =>
> s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
>
>
> exten => s,n,Record(${A_serviceline_file}.wav,0,60)
>
> I don’t call the beep file in my dialplan.
>
>
>
>
>
> *Van:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *Namens *Sammy Govind
> *Verzonden:* 05-10-2011 09:04
>
>
> *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Onderwerp:* Re: [asterisk-users] Beep file with Record
>
>
>
> How are you calling the beep.alaw from the dialplan?
>
> paste the relevant dialplan here and corresponding CLI logs.
>
>
>
> On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion <
> arjan.kr...@mobillion.nl> wrote:
>
> I placed a beep.alaw file in de directory, but I get the same result.
>
> Also I try to set the language just with two characters.
> (exten => s,n,Set(CHANNEL(language)=nl))
> And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
> beep.alaw.
> But with this also I get also the same result.
>
>
> -Oorspronkelijk bericht-
> Van: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
>
> Verzonden: 04-10-2011 17:16
>
> Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Onderwerp: Re: [asterisk-users] Beep file with Record
>
> I see two "problems" here. Problem 1 is that you are using the alaw codec,
> so it seems to me that you need this file to exist -
> /var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in
> my head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this
> is just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2
> characters, not xx/ (nl/fvdb) (feel free to correct my assumption that
> language has not been expanded beyond the 2 character limitation)?
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
> Mobillion
> Sent: Tuesday, October 04, 2011 9:43 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Beep file with Record
>
> Yes,
>
> In the code I use set the language
> exten => s,n,Set(CHANNEL(language)=nl/fvdb)
>
> So therefore I try also to place the file in the directory
> /var/lib/asterisk/sounds/nl/fvdb/
>
>
> -Oorspronkelijk bericht-
> Van: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] Namens Andrew Latham
> Verzonden: 04-10-2011 16:41
> Aan: Asterisk Users Mailing List - Non-Commercial Discussion
> Onderwerp: Re: [asterisk-users] Beep file with Record
>
> On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion <
> arjan.kr...@mobillion.nl> wrote:
> > This is my complete CLI logging
> >
> > -- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
> > "/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
> > 0") in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
> > ast_openstream
Re: [asterisk-users] Beep file with Record
Yes I already try this (only with language nl)
exten => s,n,Set(CHANNEL(language)=nl))
I also try to place the voicefile in the directory /var/lib/asterisk/sounds/
and /var/lib/asterisk/sounds/applications/ of but without any success.
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Sammy Govind
Verzonden: 05-10-2011 09:26
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
Since you've changed the language (sound directory) So just as a test change
the language back to "en" and if it goes well revert back language after the
recording.
On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion
mailto:arjan.kr...@mobillion.nl>> wrote:
CLI::
-- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
"/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60") in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
does not exist in any format
[Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory
[Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec: ast_streamfile
failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =>
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten => s,n,Record(${A_serviceline_file}.wav,0,60)
I don't call the beep file in my dialplan.
Van:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Namens Sammy Govind
Verzonden: 05-10-2011 09:04
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
How are you calling the beep.alaw from the dialplan?
paste the relevant dialplan here and corresponding CLI logs.
On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
mailto:arjan.kr...@mobillion.nl>> wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten => s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Namens Danny Nicholas
Verzonden: 04-10-2011 17:16
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two "problems" here. Problem 1 is that you are using the alaw codec, so
it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in my
head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this is
just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2 characters,
not xx/ (nl/fvdb) (feel free to correct my assumption that language has not
been expanded beyond the 2 character limitation)?
-Original Message-
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Arjan Kroon | Mobillion
Sent: Tuesday, October 04, 2011 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Beep file with Record
Yes,
In the code I use set the language
exten => s,n,Set(CHANNEL(language)=nl/fvdb)
So therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Namens Andrew Latham
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
mailto:arjan.kr...@mobillion.nl>> wrote:
> This is my complete CLI logging
>
> -- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
> "/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
> 0") in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
> ast_openstream_full: File beep does not exist in any format [Oct 4
> 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
> beep (format 0x8 (alaw)): No such file or directory [Oct 4 16:19:38]
> WARNING[13370]: app_record.c:281 record_exec: ast_streamfile failed on
> CAPI/ISDN1#02/318647615-37
>
> In de Conf file I use the following command:
> exten =>
> s,n
Re: [asterisk-users] Beep file with Record
Since you've changed the language (sound directory) So just as a test change
the language back to "en" and if it goes well revert back language after the
recording.
On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion <
arjan.kr...@mobillion.nl> wrote:
> CLI::
>
> -- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
> "/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60") in
> new stack
>
> [Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
> does not exist in any format
>
> [Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
> beep (format 0x8 (alaw)): No such file or directory
>
> [Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec:
> ast_streamfile failed on CAPI/ISDN1#02/318647615-37
>
>
>
> ** **
>
> In de Conf file I use the following command:
>
> exten =>
> s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
>
>
> exten => s,n,Record(${A_serviceline_file}.wav,0,60)
>
>
>
> I don’t call the beep file in my dialplan.
>
> ** **
>
> ** **
>
> *Van:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *Namens *Sammy Govind
> *Verzonden:* 05-10-2011 09:04
>
> *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Onderwerp:* Re: [asterisk-users] Beep file with Record
>
> ** **
>
> How are you calling the beep.alaw from the dialplan?
>
> paste the relevant dialplan here and corresponding CLI logs.
>
> ** **
>
> On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion <
> arjan.kr...@mobillion.nl> wrote:
>
> I placed a beep.alaw file in de directory, but I get the same result.
>
> Also I try to set the language just with two characters.
> (exten => s,n,Set(CHANNEL(language)=nl))
> And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
> beep.alaw.
> But with this also I get also the same result.
>
>
> -Oorspronkelijk bericht-
> Van: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
>
> Verzonden: 04-10-2011 17:16
>
> Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Onderwerp: Re: [asterisk-users] Beep file with Record
>
> I see two "problems" here. Problem 1 is that you are using the alaw codec,
> so it seems to me that you need this file to exist -
> /var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in
> my head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this
> is just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2
> characters, not xx/ (nl/fvdb) (feel free to correct my assumption that
> language has not been expanded beyond the 2 character limitation)?
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
> Mobillion
> Sent: Tuesday, October 04, 2011 9:43 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Beep file with Record
>
> Yes,
>
> In the code I use set the language
> exten => s,n,Set(CHANNEL(language)=nl/fvdb)
>
> So therefore I try also to place the file in the directory
> /var/lib/asterisk/sounds/nl/fvdb/
>
>
> -Oorspronkelijk bericht-
> Van: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] Namens Andrew Latham
> Verzonden: 04-10-2011 16:41
> Aan: Asterisk Users Mailing List - Non-Commercial Discussion
> Onderwerp: Re: [asterisk-users] Beep file with Record
>
> On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion <
> arjan.kr...@mobillion.nl> wrote:
> > This is my complete CLI logging
> >
> > -- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
> > "/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
> > 0") in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
> > ast_openstream_full: File beep does not exist in any format [Oct 4
> > 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
> > beep (format 0x8 (alaw)): No such file or directory [Oct 4 16:19:38]
> > WARNING[13370]: app_record.c:281 record_exec: ast_streamfile failed on
> > CAPI/ISDN1#02/318647615-37
> >
> > In de Conf file I use the following command:
> > exten =>
> > s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/service
> > line/${UNIQUEID) exten => s,n,Record(${A_serviceline_file}.wav,0,60)
> >
> >
> > -Oorspronkelijk bericht-
> > Van: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
> > Verzonden: 04-10-2011 16:30
> > Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Onderwerp: Re: [asterisk-users] Beep file with Record
> >
> > Usually this message is received because you did something like
> > playback(beep.gsm) or playback(beep.wav) instead of
Re: [asterisk-users] Beep file with Record
i think you can try placing the beef file in the /var/lib/asterisk/sounds
directory and not the language specific one.
and your system is calling the beep file without having it in the dialplan?
sounds strange somehow to me.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
From: arjan.kr...@mobillion.nl
To: asterisk-users@lists.digium.com
Date: Wed, 5 Oct 2011 09:20:32 +0200
Subject: Re: [asterisk-users] Beep file with Record
CLI::-- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
"/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60") in
new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full:
File beep does not exist in any format [Oct 4 16:19:38] WARNING[13370]:
file.c:950 ast_streamfile: Unable to open beep (format 0x8 (alaw)): No such
file or directory [Oct 4 16:19:38] WARNING[13370]: app_record.c:281
record_exec: ast_streamfile failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:exten =>
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten => s,n,Record(${A_serviceline_file}.wav,0,60)
I don’t call the beep file in my dialplan. Van:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Sammy Govind
Verzonden: 05-10-2011 09:04
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record How are you calling the
beep.alaw from the dialplan?paste the relevant dialplan here and corresponding
CLI logs. On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
wrote:I placed a beep.alaw file in de directory, but
I get the same result.
Also I try to set the language just with two characters.
(exten => s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny
NicholasVerzonden: 04-10-2011 17:16Aan: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two "problems" here. Problem 1 is that you are using the alaw codec, so
it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in my
head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this is
just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2 characters,
not xx/ (nl/fvdb) (feel free to correct my assumption that language has not
been expanded beyond the 2 character limitation)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
Mobillion
Sent: Tuesday, October 04, 2011 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Beep file with Record
Yes,
In the code I use set the language
exten => s,n,Set(CHANNEL(language)=nl/fvdb)
So therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Andrew Latham
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
wrote:
> This is my complete CLI logging
>
> -- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
> "/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
> 0") in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
> ast_openstream_full: File beep does not exist in any format [Oct 4
> 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
> beep (format 0x8 (alaw)): No such file or directory [Oct 4 16:19:38]
> WARNING[13370]: app_record.c:281 record_exec: ast_streamfile failed on
> CAPI/ISDN1#02/318647615-37
>
> In de Conf file I use the following command:
> exten =>
> s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/service
> line/${UNIQUEID) exten => s,n,Record(${A_serviceline_file}.wav,0,60)
>
>
> -Oorspronkelijk bericht-
> Van: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
> Verzonden: 04-10-2011 16:30
> Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Onderwerp: Re: [asterisk-users] Beep file with Record
>
> Usually this message is received because you did something like
> playback(beep.gsm) or playback(beep.wav) instead of playback(beep).
> It is
> (IMO) somewhat confusing because you have to do record(foo.gsm) but
> you have to playback using playback(foo).
Re: [asterisk-users] Beep file with Record
CLI::
-- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
"/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60") in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
does not exist in any format
[Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory
[Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec: ast_streamfile
failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =>
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten => s,n,Record(${A_serviceline_file}.wav,0,60)
I don't call the beep file in my dialplan.
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Sammy Govind
Verzonden: 05-10-2011 09:04
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
How are you calling the beep.alaw from the dialplan?
paste the relevant dialplan here and corresponding CLI logs.
On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
mailto:arjan.kr...@mobillion.nl>> wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten => s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Namens Danny Nicholas
Verzonden: 04-10-2011 17:16
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two "problems" here. Problem 1 is that you are using the alaw codec, so
it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in my
head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this is
just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2 characters,
not xx/ (nl/fvdb) (feel free to correct my assumption that language has not
been expanded beyond the 2 character limitation)?
-Original Message-
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Arjan Kroon | Mobillion
Sent: Tuesday, October 04, 2011 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Beep file with Record
Yes,
In the code I use set the language
exten => s,n,Set(CHANNEL(language)=nl/fvdb)
So therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Namens Andrew Latham
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
mailto:arjan.kr...@mobillion.nl>> wrote:
> This is my complete CLI logging
>
> -- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
> "/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
> 0") in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
> ast_openstream_full: File beep does not exist in any format [Oct 4
> 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
> beep (format 0x8 (alaw)): No such file or directory [Oct 4 16:19:38]
> WARNING[13370]: app_record.c:281 record_exec: ast_streamfile failed on
> CAPI/ISDN1#02/318647615-37
>
> In de Conf file I use the following command:
> exten =>
> s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/service
> line/${UNIQUEID) exten => s,n,Record(${A_serviceline_file}.wav,0,60)
>
>
> -Oorspronkelijk bericht-
> Van:
> asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
> Namens Danny Nicholas
> Verzonden: 04-10-2011 16:30
> Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Onderwerp: Re: [asterisk-users] Beep file with Record
>
> Usually this message is received because you did something like
> playback(beep.gsm) or playback(beep.wav) instead of playback(beep).
> It is
> (IMO) somewhat confusing because you have to do record(foo.gsm) but
> you have to playback using playback(foo).
>
> -Original Message-
> From:
> asterisk-
Re: [asterisk-users] Beep file with Record
How are you calling the beep.alaw from the dialplan?
paste the relevant dialplan here and corresponding CLI logs.
On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion <
arjan.kr...@mobillion.nl> wrote:
> I placed a beep.alaw file in de directory, but I get the same result.
>
> Also I try to set the language just with two characters.
> (exten => s,n,Set(CHANNEL(language)=nl))
> And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
> beep.alaw.
> But with this also I get also the same result.
>
> -Oorspronkelijk bericht-
> Van: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
> Verzonden: 04-10-2011 17:16
> Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Onderwerp: Re: [asterisk-users] Beep file with Record
>
> I see two "problems" here. Problem 1 is that you are using the alaw codec,
> so it seems to me that you need this file to exist -
> /var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in
> my head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this
> is just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2
> characters, not xx/ (nl/fvdb) (feel free to correct my assumption that
> language has not been expanded beyond the 2 character limitation)?
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
> Mobillion
> Sent: Tuesday, October 04, 2011 9:43 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Beep file with Record
>
> Yes,
>
> In the code I use set the language
> exten => s,n,Set(CHANNEL(language)=nl/fvdb)
>
> So therefore I try also to place the file in the directory
> /var/lib/asterisk/sounds/nl/fvdb/
>
>
> -Oorspronkelijk bericht-
> Van: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] Namens Andrew Latham
> Verzonden: 04-10-2011 16:41
> Aan: Asterisk Users Mailing List - Non-Commercial Discussion
> Onderwerp: Re: [asterisk-users] Beep file with Record
>
> On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion <
> arjan.kr...@mobillion.nl> wrote:
> > This is my complete CLI logging
> >
> > -- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
> > "/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
> > 0") in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
> > ast_openstream_full: File beep does not exist in any format [Oct 4
> > 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
> > beep (format 0x8 (alaw)): No such file or directory [Oct 4 16:19:38]
> > WARNING[13370]: app_record.c:281 record_exec: ast_streamfile failed on
> > CAPI/ISDN1#02/318647615-37
> >
> > In de Conf file I use the following command:
> > exten =>
> > s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/service
> > line/${UNIQUEID) exten => s,n,Record(${A_serviceline_file}.wav,0,60)
> >
> >
> > -Oorspronkelijk bericht-
> > Van: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
> > Verzonden: 04-10-2011 16:30
> > Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Onderwerp: Re: [asterisk-users] Beep file with Record
> >
> > Usually this message is received because you did something like
> > playback(beep.gsm) or playback(beep.wav) instead of playback(beep).
> > It is
> > (IMO) somewhat confusing because you have to do record(foo.gsm) but
> > you have to playback using playback(foo).
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan
> > Kroon | Mobillion
> > Sent: Tuesday, October 04, 2011 9:21 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] Beep file with Record
> >
> > Hi,
> >
> > I'm using the functionality Record in asterisk 1.8.5.
> > But when I want to record something I get the following error message:
> > file.c:644 ast_openstream_full: File beep does not exist in any format
> >
> > Could anybody tell me where I have to place the beep.gsm file?
> > I already tried the following directories:
> >/var/lib/asterisk/sounds/beep.gsm
> >/var/lib/asterisk/sounds/recordings/beep.gsm
> >
> > Regards,
> >
> > Arjan Kroon
>
> Beep is called from
> http://svn.asterisk.org/svn/asterisk/trunk/apps/app_record.c and it looks
> fine a first glance. Are you using the language prefix?
>
> --
> ~ Andrew "lathama" Latham lath...@gmail.com http://lathama.net ~
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
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