Re: [asterisk-users] Beep file with Record
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
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 ~
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[asterisk-users] Reinvite dialplan application [Was: OT - SIP - Toggle to autoanswer after ringing]
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
[asterisk-users] Reduce the wav file size
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
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
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Re: [asterisk-users] Asterisk (Trixbox) - VirtualBox - Linux Host
Am 04.10.11 20:40, schrieb Esteban Cacavelos: > 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.! Hello, We have several Asterisk (not Trixbox) running on OpenVZ but i guess its the same with VirtualBox. The biggest problem if you use something below 1.8 will be that you dont have access to a hardware timing source to get dahdi running, or atleast you will have to do some neat tricks to get this running. You should also think about system ressources cause asterisk will need other ressources than for example a webserver. best regards stefan -- _ -- 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
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
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Re: [asterisk-users] Database Lookup Advice
have you tried with MYSQL command? http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Tue, Oct 4, 2011 at 11:25 PM, Bryant Zimmerman wrote: > Hey all > > I wanted to get some input on what you all think is the best way to lookup > database data from asterisk dial plan. > This is a two fold question. > > 1. I am using fun_odbc to pull settings and values back and it works good > but is there a better way. I want to maintain performance and simplicity as > much as possible. > > 2. func_odbc does not appear to allow for reading multiple records in a > return set. Is there a way around this or is there a better method. > > I understand I can always us a mono, pearl, php, lua or some other kind of > script to handel some of this but I am looking for better ways to do > database access using the built-in dialplan abilities first. > > I look forward to ideas and input. > > Thanks > Bryant > > -- > _ > -- 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] Delay before ringing from PSTN`s call
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] 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
--
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[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 -- _ -- 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] Database Lookup Advice
Hey all I wanted to get some input on what you all think is the best way to lookup database data from asterisk dial plan. This is a two fold question. 1. I am using fun_odbc to pull settings and values back and it works good but is there a better way. I want to maintain performance and simplicity as much as possible. 2. func_odbc does not appear to allow for reading multiple records in a return set. Is there a way around this or is there a better method. I understand I can always us a mono, pearl, php, lua or some other kind of script to handel some of this but I am looking for better ways to do database access using the built-in dialplan abilities first. I look forward to ideas and input. Thanks Bryant -- _ -- 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] 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
--
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Re: [asterisk-users] Delay before ringing from PSTN`s call
On some analogs systems caller id is sent after first ring, so
removing "callerid=asreceived"
may help
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Tue, Oct 4, 2011 at 4:38 AM, 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'
>
> --
> _
> -- 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] USA Did required
My favorite is didww.com , another one is ipcomms.net (not very prompt with their customer service) Hope this helps.. On Sat, Oct 1, 2011 at 12:51 AM, amit mehta wrote: > Hello members, > > I am looking for USA incoming DID which can be registered on softphone/IP > Phone/ Pap2 devices. > > The DID will only be required to receive inbound calls and no outbound > calls. > > Let me know your best per month prices/cost for the above. > > Regards, > Amit Mehta > > -- > _ > -- 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
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).
>
> -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?
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[asterisk-users] OT - SIP - Toggle to autoanswer after ringing
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 -- _ -- 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
Yes,
Copy past error in mail.
In the code it is correct.
sorry
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Jose P. Espinal
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On 10/04/2011 10:37 AM, Arjan Kroon | Mobillion wrote:
> exten =>
> s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
Hello Arjam,
Did you notice that there's a missing '}' around the end of the line (on
the UNIQUEID part)?
--
# Jose P. Espinal
# http://www.eSlackware.com
# IRC: Khratos @ #asterisk / -doc / -bugs
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Re: [asterisk-users] Beep file with Record
Yes,
In the code I use set the language
exten => s,n,Set(CHANNEL(language)=nl/fvdb)
So therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Andrew Latham
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
wrote:
> This is my complete CLI logging
>
> -- Executing [s@ serviceline:93] Record("CAPI/ISDN1#02/318647615-37",
> "/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60") in
> new stack
> [Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
> does not exist in any format
> [Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
> beep (format 0x8 (alaw)): No such file or directory
> [Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec:
> ast_streamfile failed on CAPI/ISDN1#02/318647615-37
>
> In de Conf file I use the following command:
> exten =>
> s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
> exten => s,n,Record(${A_serviceline_file}.wav,0,60)
>
>
> -Oorspronkelijk bericht-
> Van: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
> Verzonden: 04-10-2011 16:30
> Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Onderwerp: Re: [asterisk-users] Beep file with Record
>
> Usually this message is received because you did something like
> playback(beep.gsm) or playback(beep.wav) instead of playback(beep). It is
> (IMO) somewhat confusing because you have to do record(foo.gsm) but you have
> to playback using playback(foo).
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
> Mobillion
> Sent: Tuesday, October 04, 2011 9:21 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Beep file with Record
>
> Hi,
>
> I'm using the functionality Record in asterisk 1.8.5.
> But when I want to record something I get the following error message:
> file.c:644 ast_openstream_full: File beep does not exist in any format
>
> Could anybody tell me where I have to place the beep.gsm file?
> I already tried the following directories:
> /var/lib/asterisk/sounds/beep.gsm
> /var/lib/asterisk/sounds/recordings/beep.gsm
>
> Regards,
>
> Arjan Kroon
Beep is called from
http://svn.asterisk.org/svn/asterisk/trunk/apps/app_record.c and it
looks fine a first glance. Are you using the language prefix?
--
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Re: [asterisk-users] Beep file with Record
On 10/04/2011 10:37 AM, Arjan Kroon | Mobillion wrote:
exten =>
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
Hello Arjam,
Did you notice that there's a missing '}' around the end of the line (on
the UNIQUEID part)?
--
# Jose P. Espinal
# http://www.eSlackware.com
# IRC: Khratos @ #asterisk / -doc / -bugs
--
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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,0,60") 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)
>
>
> -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 ~
--
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Re: [asterisk-users] Beep file with Record
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,0,60") 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)
-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
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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 -- _ -- 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] 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 -- _ -- 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 hardware
On Tue, Oct 4, 2011 at 6:06 AM, Tzafrir Cohen wrote: > On Fri, Sep 30, 2011 at 10:50:12AM -0400, Adam Moffett wrote: >> Is there any reason not to run Asterisk on an Intel Atom board? > > Only if it's not strong enough. Note that "Atom" may mean some different > things. So consider taking various reports with a few grains of salt. > > -- > Tzafrir Cohen > icq#16849755 jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir Like Tzafrir says keep an eye out for what "Atom" is. The 1.6-1.8 ghz processor is powerful enough for simple servers but some of the supporting chipsets and hardware may not be. My personal suggestion is the http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE-HF-D525.cfm which also has an IPMI onboard. -- ~ Andrew "lathama" Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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] Lag with Call Transfer (Patching)
Hi, Using Asterisk 1.6.2.13 We are now starting to use *call transfer (patching) function.* Call flow is as follows: --- John Calls me and requests him to be connected to Nancy. I place John's call on Hold I dial Nancy and speak with her about John I then patch the call between John and Nancy My line is free for the next call while John & Nancy continue their conversation In this scenario, my conversation with John & Nancy is perfect. No problems. But both John & Nancy report that the conversation between them (after I patched them) has a lag of about 4-5 seconds. I am not able to understand why this is happening. I am using GrandStream GXP280 IP Phone. Pls suggest what may be the problem area and how should I resolve this. Thank you. Best regards, Sans -- _ -- 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] rtp.conf and Asterisk as a sip agent/client
On 04/10/11 10:21, � wrote: Am 04.10.2011 10:33, schrieb Sebastian Arcus: Hello list, I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to sipgate.co.uk as a sip agent/client (with "register =>" statement in sip.conf). If I restrict the number of ports used in rtp.conf (to 1-10005 for example) - will that affect the sip sessions to sipgate.co.uk as well - or only those sessions where Asterisk acts as a sip proxy/server? Many thanks, Sebastian Hi Sebastian, it also affects the sip/rtp sessions to sipgate.co.uk. best regards, Ruben Thanks Ruben -- _ -- 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] rtp.conf and Asterisk as a sip agent/client
Am 04.10.2011 10:33, schrieb Sebastian Arcus: > Hello list, > > I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to > sipgate.co.uk as a sip agent/client (with "register =>" statement in > sip.conf). > > If I restrict the number of ports used in rtp.conf (to 1-10005 for > example) - will that affect the sip sessions to sipgate.co.uk as well - > or only those sessions where Asterisk acts as a sip proxy/server? > > Many thanks, > > Sebastian Hi Sebastian, it also affects the sip/rtp sessions to sipgate.co.uk. best regards, Ruben -- _ -- 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 hardware
On Fri, Sep 30, 2011 at 10:50:12AM -0400, Adam Moffett wrote: > Is there any reason not to run Asterisk on an Intel Atom board? Only if it's not strong enough. Note that "Atom" may mean some different things. So consider taking various reports with a few grains of salt. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] rtp.conf and Asterisk as a sip agent/client
Hello list, I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to sipgate.co.uk as a sip agent/client (with "register =>" statement in sip.conf). If I restrict the number of ports used in rtp.conf (to 1-10005 for example) - will that affect the sip sessions to sipgate.co.uk as well - or only those sessions where Asterisk acts as a sip proxy/server? Many thanks, Sebastian -- _ -- 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
