[asterisk-users] Tracking Music-on-Hold on call queues
Hi all, we have a little tool that tracks Music-on-Hold events for call queues by listening to AMI events. This is quite useful for reporting so, as the tool is free to use and does not depend on our QueueMetrics Call Center suite, I thought I'd announce it in here as well. If anyone is interested, you can find a post here: https://www.queuemetrics.com/blog/2017/03/22/TrackingMOH/?lid=A002 Comments welcome :) l. -- Loway - home of QueueMetrics - http://queuemetrics.com WombatDialer - next generation predictive dialer - http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic Music on Hold
Thanks Carlos, I have created the table and changed the extconfig to : musiconhold = mysql,asterisk,musiconhold It works fine. Yaron On Mon, Feb 23, 2015 at 6:57 PM, Carlos Chavez cur...@telecomabmex.com wrote: On 2/23/15 3:03 AM, Yaron Nachum wrote: Hello everyone, I am trying to activate Music On Hold using DB on Asterisk 13. It works fine but in order to use new Music On hold definitions I have to reload the moh module. - The following is my configuration in extconfig.conf - I added the following line: musiconhold.conf = mysql,asterisk,bit_ast_config - The following is the table in the database: mysql select * from bit_ast_config; +++-++-- -+---+---+--+ | id | cat_metric | var_metric | commented | filename | category | var_name | var_val | +++-++-- -+---+---+--+ | 2 | 0 | 0 | 0 | musiconhold.conf | yaron | directory | moh | | 3 | 0 | 0 | 0 | musiconhold.conf | yaron | mode | files | | 10 | 0 | 0 | 0| musiconhold.conf | yaron1| directory | moh | | 11 | 0 | 0 | 0| musiconhold.conf | yaron1| mode | files| +++-++-- -+---+---+--+ Is there a way to do automatically add new moh definitions without reloading the moh module? Thanks, Yaron. You actually want to use the realtime database and not the static. With the realtime database all changes will take effect immediately. The following link explains the difference between realtime and static: https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration Here is the structure I use: CREATE TABLE `musiconhold` ( `name` varchar(80) COLLATE utf8_unicode_ci NOT NULL, `directory` varchar(255) COLLATE utf8_unicode_ci NOT NULL DEFAULT '', `application` varchar(255) COLLATE utf8_unicode_ci NOT NULL DEFAULT '', `mode` varchar(80) COLLATE utf8_unicode_ci NOT NULL DEFAULT '', `digit` char(1) COLLATE utf8_unicode_ci NOT NULL DEFAULT '', `sort` varchar(16) COLLATE utf8_unicode_ci NOT NULL DEFAULT '', `format` varchar(16) COLLATE utf8_unicode_ci NOT NULL DEFAULT '', PRIMARY KEY (`name`) ) ENGINE=MyISAM DEFAULT CHARSET=utf8 COLLATE=utf8_unicode_ci; -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- 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] Dynamic Music on Hold
Hello everyone, I am trying to activate Music On Hold using DB on Asterisk 13. It works fine but in order to use new Music On hold definitions I have to reload the moh module. - The following is my configuration in extconfig.conf - I added the following line: musiconhold.conf = mysql,asterisk,bit_ast_config - The following is the table in the database: mysql select * from bit_ast_config; +++-++---+---+---+--+ | id | cat_metric | var_metric | commented | filename| category | var_name | var_val | +++-++---+---+---+--+ | 2 | 0 | 0 | 0 | musiconhold.conf | yaron | directory | moh | | 3 | 0 | 0 | 0 | musiconhold.conf | yaron | mode | files | | 10 | 0 | 0 | 0 | musiconhold.conf | yaron1| directory | moh | | 11 | 0 | 0 | 0 | musiconhold.conf | yaron1| mode | files | +++-++---+---+---+--+ Is there a way to do automatically add new moh definitions without reloading the moh module? Thanks, Yaron. -- _ -- 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] Setting Music on Hold with the Manager Interface
On Sun, Oct 26, 2014 at 10:42 PM, Todd R. tjrl...@live.com wrote: Does anyone know how to set the music on hold class with the Manager Interface in 1.8? Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within the same login, socket session. Just trying to add this additional task. This is from PHP as you may have recognized. I have also tried surrounding musicclass with CHANNEL() but that didn't work and didn't seem right anyhow since it already knows it's a channel variable. Thanks in advance for any help on this. # Set the Music on Hold fputs($socket2, Action: Setvar\r\n); fputs($socket2, Channel: .$channel.\r\n); fputs($socket2, Variable: musicclass\r\n); fputs($socket2, Value: .$mohclass.\r\n); Use the CHANNEL function: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL Action: Setvar Channel: (your channel name here) Variable: CHANNEL(musicclass) Value: (your MoH class here) -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
Re: [asterisk-users] Setting Music on Hold with the Manager Interface
Thanks Matt. I tried that already, no luck. Still, I get blank nothingness instead of MOH. I will try again just to be sure I didn't miss something. I have also tried surrounding musicclass with CHANNEL() but that didn't work and didn't seem right anyhow since it already knows it's a channel variable. Date: Mon, 27 Oct 2014 08:51:42 -0500 Subject: Re: [asterisk-users] Setting Music on Hold with the Manager Interface From: mjor...@digium.com To: tjrl...@live.com; asterisk-users@lists.digium.com On Sun, Oct 26, 2014 at 10:42 PM, Todd R. tjrl...@live.com wrote: Does anyone know how to set the music on hold class with the Manager Interface in 1.8? Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within the same login, socket session. Just trying to add this additional task. This is from PHP as you may have recognized. I have also tried surrounding musicclass with CHANNEL() but that didn't work and didn't seem right anyhow since it already knows it's a channel variable. Thanks in advance for any help on this.# Set the Music on Hold fputs($socket2, Action: Setvar\r\n); fputs($socket2, Channel: .$channel.\r\n); fputs($socket2, Variable: musicclass\r\n); fputs($socket2, Value: .$mohclass.\r\n); Use the CHANNEL function: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL Action: SetvarChannel: (your channel name here)Variable: CHANNEL(musicclass)Value: (your MoH class here) -- Matthew Jordan Digium, Inc. | Engineering Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USACheck us out at: http://digium.com http://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] Setting Music on Hold with the Manager Interface
Does anyone know how to set the music on hold class with the Manager Interface in 1.8? Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within the same login, socket session. Just trying to add this additional task. This is from PHP as you may have recognized. I have also tried surrounding musicclass with CHANNEL() but that didn't work and didn't seem right anyhow since it already knows it's a channel variable. Thanks in advance for any help on this.# Set the Music on Hold fputs($socket2, Action: Setvar\r\n); fputs($socket2, Channel: .$channel.\r\n); fputs($socket2, Variable: musicclass\r\n); fputs($socket2, Value: .$mohclass.\r\n); -- _ -- 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] asterisk music on hold recommendations
Hi all, I'm wondering what the recommendations are for using music on hold on asterisk. As far as I understood from various pages on the web and a response from the IRC channel, I am to avoid using mp3 files because of licensing and transcoding issues. correct? I am currently using asterisk 1.8 with the mpg123 processes (mode=mp3 or mode=quietmp3 in the conf file). This means that there is one single shared stream of moh for all channels that are using the same class of moh. If I were to start using wav files (mode=files), is there a way to have the same kind of shared stream of moh to reduce the load on the machine in the case where a lot calls are on hold? Is it even worth it to try reducing the load (maybe asterisk handles playing wav files very efficiently and the extra load generated by it is negligible)? I am looking to upgrade to asterisk 11 in the future. Is any of this different for that version? Thanks for any responses! Frederic -- _ -- 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 music on hold recommendations
On Tue, Apr 23, 2013 at 02:30:24PM +0200, Frederic Van Espen wrote: Hi all, I'm wondering what the recommendations are for using music on hold on asterisk. As far as I understood from various pages on the web and a response from the IRC channel, I am to avoid using mp3 files because of licensing and transcoding issues. correct? Short version: Not really. But just use the built in The earliest moh support Asterisk had was playing of MP3 files (or piping the output of an external command). Only later on native MoH was developed - playing any file Asterisk could play. At the time Digium licensed a set of mp3 files from FreePlay Music that could be freely used as MoH files with Asterisk. Later on a certain more subtle licensing issue came up and Digium chose to stop distributing those MoH files with Asterisk. They were replaced with a set of five files which are: * Longer * Better licensed (CC-BY-SA 3.0) * Available in all the required formats So the licensing issues in question are: * MP3 is patent-encumbered and some Linux distribution keep out even MP3 playing code (other only remove MP3 encoding code). * If you don't intend to play it to a MP3 channel, why waste CPU resources on transcoding it? The newer files are available in more convinient formats. IIRC the license of the FPM ones prevented Digium from distributing modified copies. I am currently using asterisk 1.8 with the mpg123 processes (mode=mp3 or mode=quietmp3 in the conf file). If you use that mode, you're probably doing something wrong following an ancient guide. This means that there is one single shared stream of moh for all channels that are using the same class of moh. If I were to start using wav files (mode=files), is there a way to have the same kind of shared stream of moh to reduce the load on the machine in the case where a lot calls are on hold? Is it even worth it to try reducing the load (maybe asterisk handles playing wav files very efficiently and the extra load generated by it is negligible)? I am looking to upgrade to asterisk 11 in the future. Is any of this different for that version? -- 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
Re: [asterisk-users] asterisk music on hold recommendations
On 04/23/2013 03:12 PM, Tzafrir Cohen wrote: If you use that mode, you're probably doing something wrong following an ancient guide. Well, these modes are the ones documented in the sample conf files that came with asterisk 1.8.13.0: snip ; valid mode options: ; files -- read files from a directory in any Asterisk supported ; media format ; quietmp3 -- default ; mp3 -- loud ; mp3nb -- unbuffered ; quietmp3nb -- quiet unbuffered ; custom -- run a custom application (See examples below) snip Seems I did miss something from the sample file before though: snip ;cachertclasses=yes ; use 1 instance of moh class for all users who are using it, ; decrease consumable cpu cycles and memory ; disabled by default snip That seems to answer my other question. Anyone got any experience using this? Thanks! -- _ -- 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] Disabling Music On Hold
On 01/28/2011 Kevin P. Fleming wrote: Loading or not loading a MOH provider is not going to change Asterisk's behavior regarding hold/unhold of endpoints; if you want Asterisk to pass through hold/unhold indications over SIP, unfortunately it can't do that yet... although most of the code has been written, it has not quite been finished. Hello Kevin, thanks a lot for this information. Is it planned that this feature will get finished at some point? Would it be appropriate to write up an issue so that this is being tracked? PS: on the wiki page http://www.voip-info.org/wiki/index.php?page_id=400 under canreinvite there is a small statement that refers to this case: If a reINVITE is needed to switch a media stream to inactive (when placed on hold) or to T.38, it will still be done, regardless of this setting! regards Urs-- _ -- 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] Disabling Music On Hold
On 01/31/2011 02:06 AM, Urs Buob wrote: On 01/28/2011 Kevin P. Fleming wrote: Loading or not loading a MOH provider is not going to change Asterisk's behavior regarding hold/unhold of endpoints; if you want Asterisk to pass through hold/unhold indications over SIP, unfortunately it can't do that yet... although most of the code has been written, it has not quite been finished. Hello Kevin, thanks a lot for this information. Is it planned that this feature will get finished at some point? I am not aware of this work being on anyone's plans, but of course it's not possible to know what community members will be working on in the future. There has not been any significant amount of demand for this feature to be finished from the user community to my knowledge. Would it be appropriate to write up an issue so that this is being tracked? Not really, no, because it's not a bug, and 'feature request' issues without patches attached to implement the feature are usually closed rather quickly (since there are over 700 open issues already, they wouldn't be likely to get attention very soon). PS: on the wiki page http://www.voip-info.org/wiki/index.php?page_id=400under canreinvite there is a small statement that refers to this case: If a reINVITE is needed to switch a media stream to inactive (when placed on hold) or to T.38, it will still be done, regardless of this setting! Well, that statement is not false, even if there isn't currently a way to cause that part of the behavior to be triggered :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] Disabling Music On Hold
Hello,
I have been trying to completely disable music on hold on my asterisk
system. When a call is put on hold I do not want any music on hold, but I
would like the remote user to get informed of this event (depending on the
technology e.g. with a SIP reinvite and an SDP indicating the call is on
hold).
I have searched and tried out various approaches, but when putting the
call on hold by a SIP user, I always get an indication that asterisk tries
to play music on hold. The remote side does not get a re-invite.
What I have tried so far:
- no musiconhold.conf in the hope that lack of the configuration file
disables moh
- a musiconhold.conf where everything is commented out
- modules.conf with 'unload = res_musiconhold.so'
When I start asterisk, it indicates that it disables music on hold:
[Jan 28 10:15:02] WARNING[31052]: res_musiconhold.c:1784 load_module: No
music on hold classes configured, disabling music on hold.
== Registered application 'MusicOnHold'
== Registered application 'WaitMusicOnHold'
== Registered application 'SetMusicOnHold'
== Registered application 'StartMusicOnHold'
== Registered application 'StopMusicOnHold'
res_musiconhold.so = (Music On Hold Resource)
However, when I set up a sip call between two sip phones and one end puts
the call on hold, then I always get the following message and the remote
side is not informed that the call is on hold:
-- Executing [s@macro-stddial:2] Dial(SIP/-, SIP/)
in new stack
== Using SIP RTP CoS mark 5
-- Called
-- SIP/-0001 is ringing
-- SIP/-0001 answered SIP/-
-- Native bridging SIP/- and SIP/-0001
later when the call is put on hold:
-- Music class default requested but no musiconhold loaded.
Can anybody give me any pointers or tell me how to disable moh completely
and send re-invites for call hold?
thanks for any help
Urs
My easiest configuration with Asterisk 1.6.2.7:
modules.conf
--
[modules]
autoload=yes
; res_phoneprov requires func_strings.so to be loaded:
preload = func_strings.so
noload = pbx_gtkconsole.so
noload = res_musiconhold.so
extensions.conf:
---
[general]
[default]
;SIP extensions
exten = _,1,Macro(stddial,SIP/${EXTEN})
[macro-stddial]
; ${ARG1} - What to dial
exten = s,1,Answer()
exten = s,n,Dial(${ARG1})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Hangup
exten = _s-.,1,NoOp
sip.conf:
[general]
language=en ; configured default language
dtmfmode=rfc2833; default dtmfmode for sending DTMF
(Dual-tone multi-frequency)
directrtpsetup=no ; Disable the new experimental direct
RTP setup
allowtransfer=yes ; enable all transfers for peers and
users
match_auth_username=yes ; use 'authentication username'
instead of 'username for authentication' (if available)
session-timers=originate; Request and run session-timers
always
session-expires=3600; maximum session refresh interval
session-minse=600 ; minimum session refresh interval
session-refresher=uas ; session refresher is
user-agent-server
;allowguest=no ; Allow or reject guest calls (default
is yes)
notifyhold = yes ; Notify subscriptions on HOLD state
(default: no)
udpbindaddr=0.0.0.0:5060; Servers IP address (all) to bind UDP
listen socket to
srvlookup=yes ; enable DNS SRV lookups on outbound
calls
[allusers](!)
context=default
type=friend ; All options are possible
qualify=2000; no, 2000=2 sec to wait for reply
before remote party is considered unreachable
;qualifyfreq=60 ; Qualification: How often to check in
seconds
canreinvite=yes ; certain devices do not like change
of RTP source/destination during call
[](allusers)
host=dynamic ; the device needs to register
secret = 1234
[](allusers)
host=dynamic ; the device needs to register
secret = 1234
[](allusers)
host=dynamic ; the device needs to register
secret = 1234
--
_
-- 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] Disabling Music On Hold
On 11-01-28 07:37 AM, Urs Buob wrote: modules.conf -- [modules] autoload=yes ; res_phoneprov requires func_strings.so to be loaded: preload = func_strings.so noload = pbx_gtkconsole.so noload = res_musiconhold.so This is the correct method. But you are saying even if you stop and start Asterisk res_musiconhold.so is still loads? I just tested with the latest 1.6.2 branch with the same settings, MOH was not loaded. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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
Re: [asterisk-users] Disabling Music On Hold
On 11-01-28 07:37 AM, Urs Buob wrote: modules.conf -- [modules] autoload=yes ; res_phoneprov requires func_strings.so to be loaded: preload = func_strings.so noload = pbx_gtkconsole.so noload = res_musiconhold.so This is the correct method. But you are saying even if you stop and start Asterisk res_musiconhold.so is still loads? I just tested with the latest 1.6.2 branch with the same settings, MOH was not loaded. Well, I did not say that MOH get's loaded. I just say that asterisk is still trying to play MOH and does NOT inform the remote side of the hold status. Actually the error message that the CLI shows when I put the call on hold also indicates that MOH is not loaded. -- Music class default requested but no musiconhold loaded. So, the problem is not that MOH is loaded, but that asterisk still tries to invoke MOH (triggering the error message) and that there is no re-invite to the remote SIP user indicating that the call is on hold. My main goal is to have a clean hold functionality with re-invites that asterisk sends out. (RTP stream goes via asterisk and not directly between the SIP clients) regards Urs-- _ -- 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] Disabling Music On Hold
On 01/28/2011 11:02 AM, Urs Buob wrote: On 11-01-28 07:37 AM, Urs Buob wrote: modules.conf -- [modules] autoload=yes ; res_phoneprov requires func_strings.so to be loaded: preload = func_strings.so noload = pbx_gtkconsole.so noload = res_musiconhold.so This is the correct method. But you are saying even if you stop and start Asterisk res_musiconhold.so is still loads? I just tested with the latest 1.6.2 branch with the same settings, MOH was not loaded. Well, I did not say that MOH get's loaded. I just say that asterisk is still trying to play MOH and does NOT inform the remote side of the hold status. Actually the error message that the CLI shows when I put the call on hold also indicates that MOH is not loaded. -- Music class default requested but no musiconhold loaded. This is not an error message, it's informational. So, the problem is not that MOH is loaded, but that asterisk still tries to invoke MOH (triggering the error message) and that there is no re-invite to the remote SIP user indicating that the call is on hold. My main goal is to have a clean hold functionality with re-invites that asterisk sends out. (RTP stream goes via asterisk and not directly between the SIP clients) Loading or not loading a MOH provider is not going to change Asterisk's behavior regarding hold/unhold of endpoints; if you want Asterisk to pass through hold/unhold indications over SIP, unfortunately it can't do that yet... although most of the code has been written, it has not quite been finished. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] start music on hold coredump
Hi the following is message,Any advice appreciated, thank you. (gdb) bt #0 0x00429410 in __kernel_vsyscall () #1 0x00bead80 in raise () from /lib/libc.so.6 #2 0x00bec691 in abort () from /lib/libc.so.6 #3 0x00c2324b in __libc_message () from /lib/libc.so.6 #4 0x00c2b883 in _int_malloc () from /lib/libc.so.6 #5 0x00c2d3ab in malloc () from /lib/libc.so.6 #6 0x00c21ff3 in vasprintf () from /lib/libc.so.6 #7 0x00c07efe in asprintf () from /lib/libc.so.6 #8 0x080a059f in build_filename (filename=0xb77b02d0 /var/lib/asterisk/mohmp3/wq//cn/wq5, ext=0xb77af1a0 WAV) at file.c:276 #9 0x080a18f5 in ast_filehelper (filename=0xb77b02d0 /var/lib/asterisk/mohmp3/wq//cn/wq5, arg2=0x0, fmt=0x0, action=ACTION_EXISTS) at file.c:445 #10 0x080a250b in fileexists_core (filename=0x9140ce0 /var/lib/asterisk/mohmp3/wq//wq5, fmt=0x0, preflang=0x2a7e Address 0x2a7e out of bounds, buf=0xb77b02d0 /var/lib/asterisk/mohmp3/wq//cn/wq5, buflen=38) at file.c:601 #11 0x080a28bd in ast_openstream_full (chan=0x9079408, filename=0x9140ce0 /var/lib/asterisk/mohmp3/wq//wq5, preflang=0x8fff593 cn, asis=1) at file.c:709 #12 0x001cb7bc in moh_files_generator (chan=0x9079408, data=0xb7c440b0, len=160, samples=160) at res_musiconhold.c:264 #13 0x080844a3 in ast_read_generator_actions (chan=0x9079408, f=0x8f8b0dc) at channel.c:1925 #14 0x08086b85 in __ast_read (chan=0x9079408, dropaudio=0) at channel.c:2315 #15 0x08073c99 in autoservice_run (ign=0x0) at autoservice.c:114 #16 0x0810072b in dummy_start (data=0x90e37f0) at utils.c:895 ---Type return to continue, or q return to quit--- #17 0x00d1349b in start_thread () from /lib/libpthread.so.0 #18 0x00c9342e in clone () from /lib/libc.so.6 -- Best regards! jordan pan -- _ -- 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] realtime music on hold
Hello list ?!
Is there anyone that can point me to the documentation please ?
I have added a new table like on
http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
With the following values :
`musiconhold` (`name`, `directory`, `application`, `mode`, `digit`,
`sort`, `format`) VALUES
('testmoh', '/var/lib/asterisk/moh/123456', '', 'files', '', '', '');
Asterisk does not see the new MusicOnHold :
asterisk*CLI module reload extconfig
[Jul 16 17:11:09] == Parsing '/etc/asterisk/extconfig.conf': [Jul 16
17:11:09] Found
[Jul 16 17:11:09] == Binding voicemail to mysql/asterisk/voicemail_users
[Jul 16 17:11:09] == Binding sipusers to mysql/asterisk/sip_buddies
[Jul 16 17:11:09] == Binding sippeers to mysql/asterisk/sip_buddies
[Jul 16 17:11:09] == Binding queues to mysql/asterisk/queues
[Jul 16 17:11:09] == Binding queue_members to mysql/asterisk/queue_members
[Jul 16 17:11:09] == Binding meetme to mysql/asterisk/conference
*[Jul 16 17:11:09] == Binding musiconhold to mysql/asterisk/musiconhold*
asterisk*CLI moh reload
[Jul 16 17:12:53] == Parsing '/etc/asterisk/musiconhold.conf': [Jul 16
17:12:53] Found
asterisk*CLI moh show classes
Class: default
Mode: files
Directory: /var/lib/asterisk/moh
My SIPuser test2 has a value testmoh for the field 'musiconhold', but
when I put a caller on hold, the musiconhold class stays default.
[Jul 16 17:17:27] -- Called test6
[Jul 16 17:17:27] -- SIP/test6-000e is ringing
[Jul 16 17:17:30] -- SIP/test6-000e answered SIP/test2-000d
*[Jul 16 17:17:33] -- Started music on hold, class 'default', on
SIP/test6-000e*
Kind regards,
Jonas.
--
_
-- 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] realtime music on hold
It looks like theres no much information out there about using realtime moh
Have you tried making an extension that goes to MusicOnHold(testmoh)
On Fri, Jul 16, 2010 at 8:21 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello list ?!
Is there anyone that can point me to the documentation please ?
I have added a new table like on
http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
With the following values :
`musiconhold` (`name`, `directory`, `application`, `mode`, `digit`, `sort`,
`format`) VALUES
('testmoh', '/var/lib/asterisk/moh/123456', '', 'files', '', '', '');
Asterisk does not see the new MusicOnHold :
asterisk*CLI module reload extconfig
[Jul 16 17:11:09] == Parsing '/etc/asterisk/extconfig.conf': [Jul 16
17:11:09] Found
[Jul 16 17:11:09] == Binding voicemail to mysql/asterisk/voicemail_users
[Jul 16 17:11:09] == Binding sipusers to mysql/asterisk/sip_buddies
[Jul 16 17:11:09] == Binding sippeers to mysql/asterisk/sip_buddies
[Jul 16 17:11:09] == Binding queues to mysql/asterisk/queues
[Jul 16 17:11:09] == Binding queue_members to mysql/asterisk/queue_members
[Jul 16 17:11:09] == Binding meetme to mysql/asterisk/conference
[Jul 16 17:11:09] == Binding musiconhold to mysql/asterisk/musiconhold
asterisk*CLI moh reload
[Jul 16 17:12:53] == Parsing '/etc/asterisk/musiconhold.conf': [Jul 16
17:12:53] Found
asterisk*CLI moh show classes
Class: default
Mode: files
Directory: /var/lib/asterisk/moh
My SIPuser test2 has a value testmoh for the field 'musiconhold', but when
I put a caller on hold, the musiconhold class stays default.
[Jul 16 17:17:27] -- Called test6
[Jul 16 17:17:27] -- SIP/test6-000e is ringing
[Jul 16 17:17:30] -- SIP/test6-000e answered SIP/test2-000d
[Jul 16 17:17:33] -- Started music on hold, class 'default', on
SIP/test6-000e
Kind regards,
Jonas.
--
_
-- 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] realtime music on hold
On Fri, 2010-07-16 at 09:35 -0700, Kyle Kienapfel wrote:
It looks like theres no much information out there about using realtime moh
Have you tried making an extension that goes to MusicOnHold(testmoh)
On Fri, Jul 16, 2010 at 8:21 AM, Jonas Kellens jonas.kell...@telenet.be
wrote:
Hello list ?!
Is there anyone that can point me to the documentation please ?
I have added a new table like on
http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
With the following values :
`musiconhold` (`name`, `directory`, `application`, `mode`, `digit`, `sort`,
`format`) VALUES
('testmoh', '/var/lib/asterisk/moh/123456', '', 'files', '', '', '');
Asterisk does not see the new MusicOnHold :
asterisk*CLI module reload extconfig
[Jul 16 17:11:09] == Parsing '/etc/asterisk/extconfig.conf': [Jul 16
17:11:09] Found
[Jul 16 17:11:09] == Binding voicemail to mysql/asterisk/voicemail_users
[Jul 16 17:11:09] == Binding sipusers to mysql/asterisk/sip_buddies
[Jul 16 17:11:09] == Binding sippeers to mysql/asterisk/sip_buddies
[Jul 16 17:11:09] == Binding queues to mysql/asterisk/queues
[Jul 16 17:11:09] == Binding queue_members to mysql/asterisk/queue_members
[Jul 16 17:11:09] == Binding meetme to mysql/asterisk/conference
[Jul 16 17:11:09] == Binding musiconhold to mysql/asterisk/musiconhold
asterisk*CLI moh reload
[Jul 16 17:12:53] == Parsing '/etc/asterisk/musiconhold.conf': [Jul 16
17:12:53] Found
asterisk*CLI moh show classes
Class: default
Mode: files
Directory: /var/lib/asterisk/moh
My SIPuser test2 has a value testmoh for the field 'musiconhold', but
when I put a caller on hold, the musiconhold class stays default.
[Jul 16 17:17:27] -- Called test6
[Jul 16 17:17:27] -- SIP/test6-000e is ringing
[Jul 16 17:17:30] -- SIP/test6-000e answered SIP/test2-000d
[Jul 16 17:17:33] -- Started music on hold, class 'default', on
SIP/test6-000e
Here is what I use:
CREATE TABLE `musiconhold` (
`name` varchar(80) collate utf8_unicode_ci NOT NULL,
`directory` varchar(255) collate utf8_unicode_ci NOT NULL default '',
`application` varchar(255) collate utf8_unicode_ci NOT NULL default
'',
`mode` varchar(80) collate utf8_unicode_ci NOT NULL default '',
`digit` char(1) collate utf8_unicode_ci NOT NULL default '',
`sort` varchar(16) collate utf8_unicode_ci NOT NULL default '',
`format` varchar(16) collate utf8_unicode_ci NOT NULL default '',
PRIMARY KEY (`name`)
) ENGINE=MyISAM DEFAULT CHARSET=utf8 COLLATE=utf8_unicode_ci;
SET character_set_client = @saved_cs_client;
INSERT INTO `musiconhold` VALUES
('default','/var/lib/asterisk/moh','','files','','random',''),('anuncios','/var/lib/asterisk/moh/anuncios','','files','','alpha','');
It works but it will not show anything with moh show classes or moh
show files. This is the case with most realtime modules unless there
is a solution like rtcachefriends=yes in sip.conf. There is also a
comment somewhere that you need to have at least one definition in
musiconhold.conf for realtime to work. I usually leave the file as is
and realtime works.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
signature.asc
Description: This is a digitally signed message part
--
_
-- 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] realtime music on hold
Thank you for your input. It seemed like a good approach, but it confirms that Asterisk does not see the new MusicOnHold-class : The dialplan : exten = 60,1,NoOp() exten = 60,n,MusicOnHold(testmoh) The CLI : [Jul 16 19:40:45] -- Executing [...@from-test:2] MusicOnHold(SIP/test2-000f, testmoh) in new stack [Jul 16 19:40:45] WARNING[30772]: res_musiconhold.c:666 get_mohbyname: Music on Hold class 'testmoh' not found [Jul 16 19:40:45] -- Started music on hold, class 'default', on SIP/test2-000f Kind regards, Jonas. On 07/16/2010 06:35 PM, Kyle Kienapfel wrote: It looks like theres no much information out there about using realtime moh Have you tried making an extension that goes to MusicOnHold(testmoh) On Fri, Jul 16, 2010 at 8:21 AM, Jonas Kellensjonas.kell...@telenet.be wrote: -- _ -- 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] realtime music on hold
On 07/16/2010 07:43 PM, Carlos Chavez wrote:
Here is what I use:
CREATE TABLE `musiconhold` (
`name` varchar(80) collate utf8_unicode_ci NOT NULL,
`directory` varchar(255) collate utf8_unicode_ci NOT NULL default '',
`application` varchar(255) collate utf8_unicode_ci NOT NULL default
'',
`mode` varchar(80) collate utf8_unicode_ci NOT NULL default '',
`digit` char(1) collate utf8_unicode_ci NOT NULL default '',
`sort` varchar(16) collate utf8_unicode_ci NOT NULL default '',
`format` varchar(16) collate utf8_unicode_ci NOT NULL default '',
PRIMARY KEY (`name`)
) ENGINE=MyISAM DEFAULT CHARSET=utf8 COLLATE=utf8_unicode_ci;
SET character_set_client = @saved_cs_client;
INSERT INTO `musiconhold` VALUES
('default','/var/lib/asterisk/moh','','files','','random',''),('anuncios','/var/lib/asterisk/moh/anuncios','','files','','alpha','');
It works but it will not show anything with moh show classes or moh
show files. This is the case with most realtime modules unless there
is a solution like rtcachefriends=yes in sip.conf. There is also a
comment somewhere that you need to have at least one definition in
musiconhold.conf for realtime to work. I usually leave the file as is
and realtime works.
I found the following :
/musiconhold.conf has a general section now. It has one option:
cachertclasses. If set to yes,
then moh classes found in realtime will be added to the in-memory
list. This has the advantage
of not requiring database lookups each time moh is started, but it
has the disadvantage of not
truly being realtime./
But when setting the following in musiconhold.conf :
[general]
cachertclasses=yes
[default]
mode=files
directory=/var/lib/asterisk/moh
random=yes
This is what the Asterisk CLI shows :
asterisk*CLI moh reload
[Jul 16 20:01:13] == Parsing '/etc/asterisk/musiconhold.conf': [Jul 16
20:01:13] Found
[Jul 16 20:01:13] WARNING[30819]: res_musiconhold.c:1205
load_moh_classes: A directory must be specified for class 'general'!
Using asterisk 1.4.30 by the way...
Kind regards,
Jonas.
--
_
-- 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] realtime music on hold
On Fri, 2010-07-16 at 20:06 +0200, Jonas Kellens wrote:
On 07/16/2010 07:43 PM, Carlos Chavez wrote:
Here is what I use:
CREATE TABLE `musiconhold` (
`name` varchar(80) collate utf8_unicode_ci NOT NULL,
`directory` varchar(255) collate utf8_unicode_ci NOT NULL default '',
`application` varchar(255) collate utf8_unicode_ci NOT NULL default
'',
`mode` varchar(80) collate utf8_unicode_ci NOT NULL default '',
`digit` char(1) collate utf8_unicode_ci NOT NULL default '',
`sort` varchar(16) collate utf8_unicode_ci NOT NULL default '',
`format` varchar(16) collate utf8_unicode_ci NOT NULL default '',
PRIMARY KEY (`name`)
) ENGINE=MyISAM DEFAULT CHARSET=utf8 COLLATE=utf8_unicode_ci;
SET character_set_client = @saved_cs_client;
INSERT INTO `musiconhold` VALUES
('default','/var/lib/asterisk/moh','','files','','random',''),('anuncios','/var/lib/asterisk/moh/anuncios','','files','','alpha','');
It works but it will not show anything with moh show classes or moh
show files. This is the case with most realtime modules unless there
is a solution like rtcachefriends=yes in sip.conf. There is also a
comment somewhere that you need to have at least one definition in
musiconhold.conf for realtime to work. I usually leave the file as is
and realtime works.
I found the following :
musiconhold.conf has a general section now. It has one option:
cachertclasses. If set to yes,
then moh classes found in realtime will be added to the in-memory
list. This has the advantage
of not requiring database lookups each time moh is started, but it
has the disadvantage of not
truly being realtime.
But when setting the following in musiconhold.conf :
[general]
cachertclasses=yes
[default]
mode=files
directory=/var/lib/asterisk/moh
random=yes
This is what the Asterisk CLI shows :
asterisk*CLI moh reload
[Jul 16 20:01:13] == Parsing '/etc/asterisk/musiconhold.conf': [Jul
16 20:01:13] Found
[Jul 16 20:01:13] WARNING[30819]: res_musiconhold.c:1205
load_moh_classes: A directory must be specified for class 'general'!
Using asterisk 1.4.30 by the way...
It works as advertised in 1.6.2.X, except for the cachertclasses=yes
which still does not show information in the CLI.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
signature.asc
Description: This is a digitally signed message part
--
_
-- 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] realtime music on hold
Your solution does not work for me. I've just also added the 'default'
class to my realtime DB.
[Jul 16 20:27:02] -- Called test6
[Jul 16 20:27:02] -- SIP/test6-0014 is ringing
[Jul 16 20:27:04] -- SIP/test6-0014 answered SIP/test2-0013
[Jul 16 20:27:06] -- Started music on hold, class 'default', on
SIP/test6-0014
No moh class 'mohtest'...
Reloading, restarting Asterisk does not help...
Would it only work for asterisk 1.6 ?
Kind regards,
Jonas.
On 07/16/2010 08:16 PM, Carlos Chavez wrote:
On Fri, 2010-07-16 at 20:06 +0200, Jonas Kellens wrote:
On 07/16/2010 07:43 PM, Carlos Chavez wrote:
Here is what I use:
CREATE TABLE `musiconhold` (
`name` varchar(80) collate utf8_unicode_ci NOT NULL,
`directory` varchar(255) collate utf8_unicode_ci NOT NULL default '',
`application` varchar(255) collate utf8_unicode_ci NOT NULL default
'',
`mode` varchar(80) collate utf8_unicode_ci NOT NULL default '',
`digit` char(1) collate utf8_unicode_ci NOT NULL default '',
`sort` varchar(16) collate utf8_unicode_ci NOT NULL default '',
`format` varchar(16) collate utf8_unicode_ci NOT NULL default '',
PRIMARY KEY (`name`)
) ENGINE=MyISAM DEFAULT CHARSET=utf8 COLLATE=utf8_unicode_ci;
SET character_set_client = @saved_cs_client;
INSERT INTO `musiconhold` VALUES
('default','/var/lib/asterisk/moh','','files','','random',''),('anuncios','/var/lib/asterisk/moh/anuncios','','files','','alpha','');
It works but it will not show anything with moh show classes or moh
show files. This is the case with most realtime modules unless there
is a solution like rtcachefriends=yes in sip.conf. There is also a
comment somewhere that you need to have at least one definition in
musiconhold.conf for realtime to work. I usually leave the file as is
and realtime works.
I found the following :
musiconhold.conf has a general section now. It has one option:
cachertclasses. If set to yes,
then moh classes found in realtime will be added to the in-memory
list. This has the advantage
of not requiring database lookups each time moh is started, but it
has the disadvantage of not
truly being realtime.
But when setting the following in musiconhold.conf :
[general]
cachertclasses=yes
[default]
mode=files
directory=/var/lib/asterisk/moh
random=yes
This is what the Asterisk CLI shows :
asterisk*CLI moh reload
[Jul 16 20:01:13] == Parsing '/etc/asterisk/musiconhold.conf': [Jul
16 20:01:13] Found
[Jul 16 20:01:13] WARNING[30819]: res_musiconhold.c:1205
load_moh_classes: A directory must be specified for class 'general'!
Using asterisk 1.4.30 by the way...
It works as advertised in 1.6.2.X, except for the cachertclasses=yes
which still does not show information in the CLI.
--
_
-- 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] realtime music on hold
Hello, has anybody an idea or experience with this realtime moh ? Jonas. On 07/14/2010 08:53 PM, Jonas Kellens wrote: Hello list, using asterisk 1.4.30. When setting up the MySQL table 'musiconhold' as described in http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf , what is the meaning of the fields : `*digit*` char(1) NOT NULL default '', `*sort*` varchar(16) NOT NULL default '', and what are there default values ?! What is the default value of : `*format*` varchar(16) NOT NULL default '', Kind regards, Jonas. -- _ -- 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] realtime music on hold
Hello list, using asterisk 1.4.30. When setting up the MySQL table 'musiconhold' as described in http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf , what is the meaning of the fields : `*digit*` char(1) NOT NULL default '', `*sort*` varchar(16) NOT NULL default '', and what are there default values ?! What is the default value of : `*format*` varchar(16) NOT NULL default '', Kind regards, Jonas. -- _ -- 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] FW: Music on Hold problema
Please, I need help with this... Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 18 Jun 2010 15:12:25 + Subject: Re: [asterisk-users] Music on Hold problema The list of /var/lib/asterisk/mohmp3 is: -rw-rw 4 asterisk asterisk 184 Oct 19 2009 LICENSE-asterisk-moh-freeplay-wav -rw-rw-r-- 4 asterisk asterisk 882748 Oct 19 2009 QuajiroPromo.sln -rw-rw-r-- 4 asterisk asterisk 834682 Oct 19 2009 TristeAlegriaPromo.sln -rw-rw 4 asterisk asterisk 1939794 Oct 19 2009 fpm-calm-river.wav -rw-rw 4 asterisk asterisk 2582196 Oct 19 2009 fpm-sunshine.wav -rw-rw 4 asterisk asterisk 2217318 Oct 19 2009 fpm-world-mix.wav And the musiconhold.conf is: [default] mode=files directory=/var/lib/asterisk/mohmp3 random=yes [none] mode=files directory=/dev/null Thanks, Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 18 Jun 2010 09:26:16 -0500 Subject: Re: [asterisk-users] Music on Hold problema Post the /var/lib/asterisk/mohmp3 listing and musiconhold.conf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, June 18, 2010 9:18 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold problema Any ideas, please? Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 19:54:30 + Subject: Re: [asterisk-users] Music on Hold problema I have wav files in the /var/lib/asterisk/mohmp3... Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 14:36:00 -0500 Subject: Re: [asterisk-users] Music on Hold problema I see that moh is trying sln format, then ulaw, then failing. Do you have moh files in either of these formats? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Thursday, June 17, 2010 2:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold problema Hi people, I have a problem with Music On Hold, it is stopped just after starting... This is the log: [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:1] NoOp(SIP/7PBX-08229d18, Start) in new stack [Jun 17 19:20:22] DEBUG[20784] pbx.c: Launching 'MusicOnHold' [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:2] MusicOnHold(SIP/7PBX-08229d18, ) in new stack [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format slin [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Started music on hold, class 'default', on channel 'SIP/7PBX-08229d18' [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 160 sample intervals [Jun 17 19:20:22] DEBUG[20784] channel.c: Auto-deactivating generator [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format ulaw [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Stopped music on hold on SIP/7PBX-08229d18 [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 0 sample intervals Could you help me with this? Thanks, Anahi Ludueña Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! Del Lado Oscuro de Internet protegerte puedes. ¡Entra ya en www.ayudartepodria.com! Del Lado Oscuro de Internet protegerte puedes. ¡Entra ya en www.ayudartepodria.com! _ Sé el protagonista de GQ con Messenger y Vodafone Blackberry. ¡Y gana premios! http://serviciosmoviles.es.msn.com/messenger/vodafone.aspx-- _ -- 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] FW: Music on Hold problema
The moh conf file seems good. It is the standard implementation and should have worked. Just wondering if your end devices, whether they are IP phones or softphones, are setup to listen to some different codecs than ulaw and slin? Or in your sip.conf when declaring extensions you are not putting the correct codecs in the 'allow=' declaration. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-23 10:33 AM, Anahi Ludueña a_ludu...@hotmail.com wrote: Please, I need help with this... Anahi Ludueña From: a_... Date: Fri, 18 Jun 2010 15:12:25 + Subject: Re: [asterisk-users] Music on Hold problema The list of /var/lib/asterisk/mohmp3 is: -rw... -- _ -- 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] FW: Music on Hold problema
One thing to take into account and I haven't said before, sorry... I have 2 pbx, one is connecting to the other by a SIP trunk... The first pbx has the setting which I put some days ago... the second pbx has the extensions and I'm trying to use them in the call. Everything is working, except the music on hold. Thanks, Anahi Ludueña Date: Wed, 23 Jun 2010 10:44:10 -0400 From: zisha...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FW: Music on Hold problema The moh conf file seems good. It is the standard implementation and should have worked. Just wondering if your end devices, whether they are IP phones or softphones, are setup to listen to some different codecs than ulaw and slin? Or in your sip.conf when declaring extensions you are not putting the correct codecs in the 'allow=' declaration. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-23 10:33 AM, Anahi Ludueña a_ludu...@hotmail.com wrote: Please, I need help with this... Anahi Ludueña From: a_... Date: Fri, 18 Jun 2010 15:12:25 + Subject: Re: [asterisk-users] Music on Hold problema The list of /var/lib/asterisk/mohmp3 is: -rw... -- _ -- 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 _ Los cochazos de los famosos Patrick Dempsey, Tom Cruise o Michael Douglas presumen de automóvil http://motor.es.msn.com/coches/galeria.aspx?cp-documentid=152634169-- _ -- 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] Ring + Music on Hold in the same call
Hi list, is there a way to achieve in asterisk (version 1.4.x) the behavior described below? * a caller place a call to an extension, and I want the caller hears the extension ringing for some seconds, and then hears the music on hold (or a courtesy message) _in the same call;_ * the called extension must continue to ring until answered. With the m(...) option in the Dial command (like the example below) asterisk provides only music on hold while the phone rings. exten = s,n,Dial(SIP/,30,m(default)) I can not use queues because the requirements is to have 1 call and not a lot of calls. Thanks in advance, Matteo -- _ -- 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] Ring + Music on Hold in the same call
Here is one way to do it (works in 1.4.22-1.4.30 at least) exten = s,n,Dial(SIP/,10) exten = s,n,Dial(SIP/,90,m(default)) This snippet will ring for 10 seconds with Ringing, then ring for 90 seconds or until answered with MOH. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana Sent: Thursday, June 10, 2010 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ring + Music on Hold in the same call Hi list, is there a way to achieve in asterisk (version 1.4.x) the behavior described below? * a caller place a call to an extension, and I want the caller hears the extension ringing for some seconds, and then hears the music on hold (or a courtesy message) in the same call; * the called extension must continue to ring until answered. With the m(...) option in the Dial command (like the example below) asterisk provides only music on hold while the phone rings. exten = s,n,Dial(SIP/,30,m(default)) I can not use queues because the requirements is to have 1 call and not a lot of calls. Thanks in advance, Matteo -- _ -- 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] Ring + Music on Hold in the same call
Ok Danny but with this example I have 2 calls in the called phone, and this is what I have to avoid! Regards, Matteo Il 10/06/2010 15.16, Danny Nicholas ha scritto: Here is one way to do it (works in 1.4.22-1.4.30 at least) exten = s,n,Dial(SIP/,10) exten = s,n,Dial(SIP/,90,m(default)) This snippet will ring for 10 seconds with Ringing, then ring for 90 seconds or until answered with MOH. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana Sent: Thursday, June 10, 2010 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ring + Music on Hold in the same call Hi list, is there a way to achieve in asterisk (version 1.4.x) the behavior described below? a caller place a call to an extension, and I want the caller hears the extension ringing for some seconds, and then hears the music on hold (or a courtesy message) in the same call; the called extension must continue to ring until answered. With the m(...) option in the Dial command (like the example below) asterisk provides only music on hold while the phone rings. exten = s,n,Dial(SIP/,30,m(default)) I can not use queues because the requirements is to have 1 call and not a lot of calls. Thanks in advance, Matteo -- Ing. Matteo Campana - System Engineer Mobile: +39 320 4258536 Office: +39 059 821672 Fax: +39 059 821492 Web: www.klarya.it This e-mail transmission may contain legally privileged and/or confidential information. Please do not read it if you are not the intended recipient(s). Any use, distribution, reproduction or disclosure by any other person is strictly prohibited. If you have received this e-mail in error, please notify the sender and destroy the original transmission. -- _ -- 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] Ring + Music on Hold in the same call
Not sure how this would work, but you could create a special MOH file that was 10 seconds of ringing followed by the normal MOH - I know this CAN be done, just takes a bit of trial and error. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana Sent: Thursday, June 10, 2010 8:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Ring + Music on Hold in the same call Ok Danny but with this example I have 2 calls in the called phone, and this is what I have to avoid! Regards, Matteo Il 10/06/2010 15.16, Danny Nicholas ha scritto: Here is one way to do it (works in 1.4.22-1.4.30 at least) exten = s,n,Dial(SIP/,10) exten = s,n,Dial(SIP/,90,m(default)) This snippet will ring for 10 seconds with Ringing, then ring for 90 seconds or until answered with MOH. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana Sent: Thursday, June 10, 2010 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ring + Music on Hold in the same call Hi list, is there a way to achieve in asterisk (version 1.4.x) the behavior described below? * a caller place a call to an extension, and I want the caller hears the extension ringing for some seconds, and then hears the music on hold (or a courtesy message) in the same call; * the called extension must continue to ring until answered. With the m(...) option in the Dial command (like the example below) asterisk provides only music on hold while the phone rings. exten = s,n,Dial(SIP/,30,m(default)) I can not use queues because the requirements is to have 1 call and not a lot of calls. Thanks in advance, Matteo -- Ing. Matteo Campana - System Engineer Mobile: +39 320 4258536 Office: +39 059 821672 Fax: +39 059 821492 Web: http://www.klarya.it/ www.klarya.it This e-mail transmission may contain legally privileged and/or confidential information. Please do not read it if you are not the intended recipient(s). Any use, distribution, reproduction or disclosure by any other person is strictly prohibited. If you have received this e-mail in error, please notify the sender and destroy the original transmission. image002.jpgimage001.jpg-- _ -- 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] Ring + Music on Hold in the same call
Danny Nicholas wrote: Not sure how this would work, but you could create a special MOH file that was 10 seconds of ringing followed by the normal MOH – I know this CAN be done, just takes a bit of trial and error. That's what I would suggest as well. You could use Monitor() initially to call an extension that you let ring to get the ringing sound, then you could use any of a multiple of tools to combine the ringing onto the start of MoH. Leif. -- _ -- 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] 1.6.1.10 Music On Hold
Brilliant, thanks a lot. Best regards, Örn On Tue, Nov 24, 2009 at 1:39 PM, Santiago Gimeno santiago.gim...@gmail.comwrote: Hi, I think it can be related to https://issues.asterisk.org/view.php?id=16268 Best regards, Santi 2009/11/24 Örn Arnarson o...@arnarson.net Hello again, I just tried version 1.6.1.9, and the MOH works well there. It seems to be a bug introduced in 1.6.1.10. Best regards, Örn 2009/11/23 Örn Arnarson o...@arnarson.net Hello. I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold functionality has changed (or is bugged?). I have Aastra 6757i and Aastra 6731i phones, and now when i press the MusicOnHold button / change lines on the phone, MOH no longer starts. It did this in v 1.6.0.9. The invites received are exactly the same, only 1.6.1.10 doesn't ever start MOH. Is there some configuration change I need to implement for this to work properly? Was there a conscious change in Asterisk's behavior? Best regards, Örn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1.10 Music On Hold
Hello again, I just tried version 1.6.1.9, and the MOH works well there. It seems to be a bug introduced in 1.6.1.10. Best regards, Örn 2009/11/23 Örn Arnarson o...@arnarson.net Hello. I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold functionality has changed (or is bugged?). I have Aastra 6757i and Aastra 6731i phones, and now when i press the MusicOnHold button / change lines on the phone, MOH no longer starts. It did this in v 1.6.0.9. The invites received are exactly the same, only 1.6.1.10 doesn't ever start MOH. Is there some configuration change I need to implement for this to work properly? Was there a conscious change in Asterisk's behavior? Best regards, Örn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1.10 Music On Hold
Hi, I think it can be related to https://issues.asterisk.org/view.php?id=16268 Best regards, Santi 2009/11/24 Örn Arnarson o...@arnarson.net Hello again, I just tried version 1.6.1.9, and the MOH works well there. It seems to be a bug introduced in 1.6.1.10. Best regards, Örn 2009/11/23 Örn Arnarson o...@arnarson.net Hello. I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold functionality has changed (or is bugged?). I have Aastra 6757i and Aastra 6731i phones, and now when i press the MusicOnHold button / change lines on the phone, MOH no longer starts. It did this in v 1.6.0.9. The invites received are exactly the same, only 1.6.1.10 doesn't ever start MOH. Is there some configuration change I need to implement for this to work properly? Was there a conscious change in Asterisk's behavior? Best regards, Örn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6.1.10 Music On Hold
Hello. I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold functionality has changed (or is bugged?). I have Aastra 6757i and Aastra 6731i phones, and now when i press the MusicOnHold button / change lines on the phone, MOH no longer starts. It did this in v 1.6.0.9. The invites received are exactly the same, only 1.6.1.10 doesn't ever start MOH. Is there some configuration change I need to implement for this to work properly? Was there a conscious change in Asterisk's behavior? Best regards, Örn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Local music on hold -- mohinterpret=passthrough assymetrical ?
On Fri, Apr 3, 2009 at 11:11 AM, Richard Brady rnbr...@gmail.com wrote: Exvito Did you ever make any progress on this? ...no, sorry. Never got to the perfect solution. (and in all due honesty, I can't recall the exact setup we ended up deploying) -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Local music on hold -- mohinterpret=passthrough assymetrical ?
Exvito Did you ever make any progress on this? Richard On Mon, Mar 10, 2008 at 2:38 AM, Ex Vito ex.vitor...@gmail.com wrote: Hi list, I'm planning and testing a distributed asterisk deployment throughout several sites; each will be connected to the PSTN and all of them among themselves via IAX trunks. Phones will be SIP. I guess I already solved (worked-around, actually) asterisk's codec negotiation limitations regarding local G.711 utilization vs. remote G.729 while minimizing transcoding -- a bit of dial plan tweaking via the setting of SIP_CODEC variable seems to do the trick. But I digress... (with patch in issue 4825 things would be much nicer!) Now I'm still trying to improve bandwith usage with local music on hold; that is, when sip user A1, registered to server A puts sip caller B1, registered to server B, caller B1 gets server B's music on hold -- this removes the need of streaming audio from server A to server B while B1 is on hold, which in my scenario is a good thing. I post to the list trying to get peer feedback to my initial tests. The configurations I mention are always applied to both servers A and B. 1. If I set mohinterpret=passthrough + mohsuggest=default in the [general] section of iax.conf the local music on hold never works. Results: bad - A1 calls B1, B1 puts A1 on hold, A1 gets B's music bad - A1 calls B1, A1 puts B1 on hold, B1 gets A's music bad - B1 calls A1, A1 puts B1 on hold, B1 gets A's music bad - B1 calls A1, B1 puts A1 on hold, A1 gets B's music 2. If I set mohinterpret=passthrough + mohsuggest=default in the specific peer/user (friend, actually) section I get improved results but not perfect (or, at least, as I'd like them to be). Results: good - A1 calls B1, B1 puts A1 on hold, A1 gets A's music bad - A1 calls B1, A1 puts B1 on hold, B1 gets A's music good - B1 calls A1, A1 puts B1 on hold, B1 gets B's music bad - B1 calls A1, B1 puts A1 on hold, A1 gets B's music Fortunatelly, the good cases seem to be the most plausible ones. So, in my observation, the mohinterpret=passthrough behaviour is not symmetrical; that is, the hold signalling only seems to travel one way along the IAX trunk... From the side receiving the call to the side initiating it, and not the other way around. Can anyone verify this behaviour ? Am I doing something wrong or is this expected / by design behaviour ? Should I file a bug against 1. ? Against 2. ? Extra points question: Since the calls in this case are remote, from site A to site B, the codec in use is G.729 which, as you might well know, is really awfull at supporting music since it's been designed for voice only. How would one have the RTP stream renegotiated during call to G.711 when entering music on hold (local, of course, after fixing my issues above!) and back to G.729 when back to conversation ? (ok, this probably needs patching the source !... but maybe someone has an idea or has taken a different approach at this...) :-) Thanks a lot for any feedback, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replace music-on-hold on MeetMe with ringing sound
Date: Mon, 23 Jun 2008 08:00:08 -0400 From: David Backeberg [EMAIL PROTECTED] Subject: Re: [asterisk-users] Replace music-on-hold on MeetMe with ringing sound To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 MeetMe() provides very useful tones when a caller is added to a MeetMe room. That is, if you're using the musiconhold option, the agent would hear music, immediately followed by two tones, and then they would be bridged to the client. Perhaps you're running MeetMe() with those join tones disabled? Check out the docs for MeetMe. I think it's option capital i, as in Iberia. On Mon, Jun 23, 2008 at 12:19 AM, Cosmin Prund [EMAIL PROTECTED] wrote: Hello. It's been a while since I last posted (probably because my * works just fine). I'm working on something to replace call queues in my own application-specific way and I'm using MeetMe rooms to bridge agents and clients and do other things. When an agent needs to be bridged with a client I'll first put the agent in the MeetMe room and when I have confirmation that the agent is in the MeetMe room I'll send the client to the same room. My agent gets to hear music on hold while it's the only one in the conference room (it takes 1 or 2 seconds for the client to be put in the same room). Is it possible to make the agent here ringing (or replace the music on hold with a recording of ringing)? At the moment I'm telling agents when the music stops playing you're talking to the client but that just doesn't sound right and it's a bit fiddely because music on hold is music and music has pauses. One can imediatelly tell the ringing is done but they might need a few extra seconds to realise the music has stoped. On the other hand the client has no such problem since he/she hears ringing just before they get bridged to the MeetMe room. Any ideas? Thanks! -- Cosmin Prund I have the same problem and wish as the original poster, but couldn't find any information about this. The M option for meetme (play musiconhold) doesn't seem to have any switches to change MOH class. I've also checked voip-info.org variable list to see if there was any variable I could set before entering the meetme, but nothing there either from what I could find :( Does anyone have a solution to this, other than to replace the sound files in the default MOH directory? Thanks, Best regards, Tobias ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replace music-on-hold on MeetMe with ringing sound
MeetMe() provides very useful tones when a caller is added to a MeetMe room. That is, if you're using the musiconhold option, the agent would hear music, immediately followed by two tones, and then they would be bridged to the client. Perhaps you're running MeetMe() with those join tones disabled? Check out the docs for MeetMe. I think it's option capital i, as in Iberia. On Mon, Jun 23, 2008 at 12:19 AM, Cosmin Prund [EMAIL PROTECTED] wrote: Hello. It's been a while since I last posted (probably because my * works just fine). I'm working on something to replace call queues in my own application-specific way and I'm using MeetMe rooms to bridge agents and clients and do other things. When an agent needs to be bridged with a client I'll first put the agent in the MeetMe room and when I have confirmation that the agent is in the MeetMe room I'll send the client to the same room. My agent gets to hear music on hold while it's the only one in the conference room (it takes 1 or 2 seconds for the client to be put in the same room). Is it possible to make the agent here ringing (or replace the music on hold with a recording of ringing)? At the moment I'm telling agents when the music stops playing you're talking to the client but that just doesn't sound right and it's a bit fiddely because music on hold is music and music has pauses. One can imediatelly tell the ringing is done but they might need a few extra seconds to realise the music has stoped. On the other hand the client has no such problem since he/she hears ringing just before they get bridged to the MeetMe room. Any ideas? Thanks! -- Cosmin Prund ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Replace music-on-hold on MeetMe with ringing sound
Hello. It's been a while since I last posted (probably because my * works just fine). I'm working on something to replace call queues in my own application-specific way and I'm using MeetMe rooms to bridge agents and clients and do other things. When an agent needs to be bridged with a client I'll first put the agent in the MeetMe room and when I have confirmation that the agent is in the MeetMe room I'll send the client to the same room. My agent gets to hear music on hold while it's the only one in the conference room (it takes 1 or 2 seconds for the client to be put in the same room). Is it possible to make the agent here ringing (or replace the music on hold with a recording of ringing)? At the moment I'm telling agents when the music stops playing you're talking to the client but that just doesn't sound right and it's a bit fiddely because music on hold is music and music has pauses. One can imediatelly tell the ringing is done but they might need a few extra seconds to realise the music has stoped. On the other hand the client has no such problem since he/she hears ringing just before they get bridged to the MeetMe room. Any ideas? Thanks! -- Cosmin Prund ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Customize Music On Hold
Dear all, I have Asterisk 1.4.13 with the default configuration for Music On Hold. I have this in /etc/asterisk/musiconhold.conf: [default] mode=files directory=/var/lib/asterisk/moh and in /var/lib/asterisk/moh I have the default wav files: fpm-calm-river.wav fpm-sunshine.wav fpm-world-mix.wav This way the music on hold works very good. After that I use audacity to export my own MP3 files to WAV, and finally I put them into /var/lib/asterisk/moh and delete the default fpm* wav files. But when we call any other and turn on the HOLD function, the music doesn't work. How can I customize the music on hold files ??? Special thanks. Alejandro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Customize Music On Hold
Alejandro Cabrera Obed wrote: How can I customize the music on hold files ??? http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asteriskview_comment_id=13455 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Customize Music On Hold
On Wed, 30 Apr 2008, Alejandro Cabrera Obed wrote: I have this in /etc/asterisk/musiconhold.conf: [default] mode=files directory=/var/lib/asterisk/moh After that I use audacity to export my own MP3 files to WAV, and finally I put them into /var/lib/asterisk/moh and delete the default fpm* wav files. But when we call any other and turn on the HOLD function, the music doesn't work. Use the file command to compare the files that work and the files that don't. They should be RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz. I usually forget to change the Hz. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Customize Music On Hold
On Wed, 30 Apr 2008, Doug Lytle wrote: Alejandro Cabrera Obed wrote: How can I customize the music on hold files ??? http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asteriskview_comment_id=13455 Personally, that seems a bit like using a sledge hammer when a tack hammer would do the job. Plus, I'd hate to explain to my boss that I just found an obscure bug in Asterisk and crashed it trying to create some new MOH files :) I use: sox example.whatever -c 1 -s -w -r 8000 example.wav normalize example.wav Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Customize Music On Hold
Steve Edwards wrote: On Wed, 30 Apr 2008, Doug Lytle wrote: Alejandro Cabrera Obed wrote: How can I customize the music on hold files ??? http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asteriskview_comment_id=13455 Personally, that seems a bit like using a sledge hammer when a tack hammer You've lost me on that one, under the section on how to use wav files, it states: Using WAV files Asterisk has codecs for wav (pcm), gsm, g729, g726, and wav49, all of which can be used for Playback and Background. However, Asterisk does not understand ADPCM WAV files. To convert your WAV files to a format which Asterisk can understand, use the following command: sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Local music on hold -- mohinterpret=passthrough assymetrical ?
Hi list, I'm planning and testing a distributed asterisk deployment throughout several sites; each will be connected to the PSTN and all of them among themselves via IAX trunks. Phones will be SIP. I guess I already solved (worked-around, actually) asterisk's codec negotiation limitations regarding local G.711 utilization vs. remote G.729 while minimizing transcoding -- a bit of dial plan tweaking via the setting of SIP_CODEC variable seems to do the trick. But I digress... (with patch in issue 4825 things would be much nicer!) Now I'm still trying to improve bandwith usage with local music on hold; that is, when sip user A1, registered to server A puts sip caller B1, registered to server B, caller B1 gets server B's music on hold -- this removes the need of streaming audio from server A to server B while B1 is on hold, which in my scenario is a good thing. I post to the list trying to get peer feedback to my initial tests. The configurations I mention are always applied to both servers A and B. 1. If I set mohinterpret=passthrough + mohsuggest=default in the [general] section of iax.conf the local music on hold never works. Results: bad - A1 calls B1, B1 puts A1 on hold, A1 gets B's music bad - A1 calls B1, A1 puts B1 on hold, B1 gets A's music bad - B1 calls A1, A1 puts B1 on hold, B1 gets A's music bad - B1 calls A1, B1 puts A1 on hold, A1 gets B's music 2. If I set mohinterpret=passthrough + mohsuggest=default in the specific peer/user (friend, actually) section I get improved results but not perfect (or, at least, as I'd like them to be). Results: good - A1 calls B1, B1 puts A1 on hold, A1 gets A's music bad - A1 calls B1, A1 puts B1 on hold, B1 gets A's music good - B1 calls A1, A1 puts B1 on hold, B1 gets B's music bad - B1 calls A1, B1 puts A1 on hold, A1 gets B's music Fortunatelly, the good cases seem to be the most plausible ones. So, in my observation, the mohinterpret=passthrough behaviour is not symmetrical; that is, the hold signalling only seems to travel one way along the IAX trunk... From the side receiving the call to the side initiating it, and not the other way around. Can anyone verify this behaviour ? Am I doing something wrong or is this expected / by design behaviour ? Should I file a bug against 1. ? Against 2. ? Extra points question: Since the calls in this case are remote, from site A to site B, the codec in use is G.729 which, as you might well know, is really awfull at supporting music since it's been designed for voice only. How would one have the RTP stream renegotiated during call to G.711 when entering music on hold (local, of course, after fixing my issues above!) and back to G.729 when back to conversation ? (ok, this probably needs patching the source !... but maybe someone has an idea or has taken a different approach at this...) :-) Thanks a lot for any feedback, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme music on hold - when only conference member problem
Hi, I have the following problem that when someone connects to my conference and is the only member music on hold is played just for one second or less and then stops: [Feb 1 10:38:46] -- Started music on hold, class 'default', on channel 'SIP/sip.touk.pl-0083dad0' [Feb 1 10:38:46] -- Stopped music on hold on SIP/sip.touk.pl-0083dad0 I use the following command to invoke exten = s,n,MeetMe(|cdIMps) Kind regards tomasz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme music on hold - when only conference member problem
On Fri, 1 Feb 2008, Tomasz Zieleniewski wrote: Hi, I have the following problem that when someone connects to my conference and is the only member music on hold is played just for one second or less and then stops: [Feb 1 10:38:46] -- Started music on hold, class 'default', on channel 'SIP/sip.touk.pl-0083dad0' [Feb 1 10:38:46] -- Stopped music on hold on SIP/sip.touk.pl-0083dad0 I use the following command to invoke exten = s,n,MeetMe(|cdIMps) You probably don't have a timing source. If you don't have any telephony hardware installed you'll need the ztdummy module... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme music on hold - when only conference member problem
I have, I have ztdummy module loaded in the kernel On Feb 1, 2008 11:59 AM, Gordon Henderson [EMAIL PROTECTED] wrote: On Fri, 1 Feb 2008, Tomasz Zieleniewski wrote: Hi, I have the following problem that when someone connects to my conference and is the only member music on hold is played just for one second or less and then stops: [Feb 1 10:38:46] -- Started music on hold, class 'default', on channel 'SIP/sip.touk.pl-0083dad0' [Feb 1 10:38:46] -- Stopped music on hold on SIP/sip.touk.pl-0083dad0 I use the following command to invoke exten = s,n,MeetMe(|cdIMps) You probably don't have a timing source. If you don't have any telephony hardware installed you'll need the ztdummy module... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bristuff: music on hold but no dialoptions tT defined.
Hi, Iam dialing from NT ptp to SIP provider. Sometimes Asterisk is doing music on hold but there are no options like t or T in the dial command. As an result the channel got lost and an Hangup occurs. Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card. Any solution for this? Oct 22 11:20:06 VERBOSE[911] logger.c: -- SIP/sip-08206a30 answered Zap/8-1 Oct 22 11:20:23 VERBOSE[29983] logger.c: -- Started music on hold, class 'default', on channel 'Zap/8-1' Oct 22 11:20:46 VERBOSE[29983] logger.c: -- Stopped music on hold on Zap/8-1 Oct 22 11:20:46 VERBOSE[29983] logger.c: -- Started music on hold, class 'default', on channel 'Zap/8-1' Oct 22 11:20:55 VERBOSE[911] logger.c: == Spawn extension (macro-call, s, 2) exited non-zero on 'Zap /8-1' in macro 'tmp_call' ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No music on hold on ISDN line
Hello,
I have two incoming lines connected to my Asterisk
([EMAIL PROTECTED]). One voip and one ISDN line. Both go into
incoming context.
I have a problem, that when I press Flash the client who calls does not
hear music on hold, but only on the ISDN line, on VOIP everything is ok.
[incoming]
exten = s,1,SetMusicOnHold(default)
exten = s,2,AGI(test|incoming)
exten = s,3,Dial(${INOTEL_INCOMING},${RINGTIME},Tt)
exten = s,n,Hangup
exten = h,1,DeadAGI(test|dead)
Got any ideas?
-- Executing [EMAIL PROTECTED]:3] Dial(CAPI/ISDN/717817630-2,
SIP/101SIP/102|720|Tt) in new stack -- Called 101
-- Called 102
-- SIP/102-08783000 is ringing
-- SIP/102-08783000 answered CAPI/ISDN/717817630-2
==
capi_send_detect_dtmf_req:3445:ENTRY=ISDN:PLCI=0x1400:PBX_CHAN=CAPI/ISDN/717817630-2:
== Setting up DTMF detector, flag=1
Now the flash key was pressed and there is no Starting music on
hold. Got any ideas why?
[EMAIL PROTECTED] /usr/local/etc/asterisk]# cat capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=1
txgain=1
debug=no
[ISDN]
isdnmode=msn
msn=717817630
incomingmsn=717817630
controller=0
group=1
softdtmf=off
relaxdtmf=off
accountcode=
context=incoming
holdtype=local
echocancel=no
echosquelch=no
devices=2
[EMAIL PROTECTED] /usr/local/etc/asterisk]# isdnconfig
controller 0 = {
Layer 1:
description : HFC-2BDS0 128K PCI ISDN adapter
type: passive ISDN (Basic Rate, 2xB)
channels: 0x3
serial : 0xabcd
power_save : on
dialtone: enabled
attached: yes
PH-state: F7: Activated
Layer 2:
driver_type : DRVR_D_CHANNEL
}
--
.: Jakub Głazik,
.: email jabber: zytekatnuxi.pl
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Preparing music on hold
Greetings list, I've been having a go at preparing some music on hold from CDs clients have supplied, but quality seems really rather poor over compressed channels (tried g729, GSM and Speex). I've been doing the following: sox -v 0.15 filename.wav -t raw -r 8000 -s -w -c 1 filename.sln resample -ql As I understand it, I'm reducing volume to 15% (which sounds about right volume-wise) and downsampling to 8khz, 16-bit mono. I know MoH over compressed links isn't ideal conditions and is never going to be great, but I should be able to at least make it bearable. What's the prevailing opinion on using high and low pass filters? One would assume a phone handset is expected to provide frequency response in human speech zones, and not really much outside that (certainly not the 20hz-20khz one might expect of a CD). Suggestions gratefully appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Shoutcast music-on-hold
Hello List I am currently testing, using a shoutcast server as source for MOH. Here is the command im using: /usr/bin/wget -q -O - http://listen.coolfm.dk:80/ | /usr/local/bin/madplay -d -Q -z -o raw:- --mono -R 44100 - | sox -r 44100 -w -s -t raw - -r 8000 -c 1 -t raw - resample vol 0.10 I know that the normal examples, only shows using madplay without sox, but the quality is s bad when I do this, compared to using SoX to do the samplerate conversion. My problem is, that everytime somebody hangs up, and nobody is using the MOH, it seems as though it stops reading data from the shoutcast server. This results in the music re-buffering from the shoutcast server, which skips the music, and in this scenario results in a re-connect to the shoutcast server. Anybody know of a solution for this? Jon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Native music on hold not playing on incoming calls
Hi, I'm trying to make native music on hold work on my Asterisk 1.2.9.1 server with a Sangoma PRI card. If I use a IAX phone connected to the PBX, I hear the music, but if I make a call from outside I hear nothing even if Asterisk console says music has started... it seems something related to zapata.conf but I cannot understand what's wrong. I also put musiconhold=native for every channel inside zapata.conf without success. Is there anybody who can help me, please? TIA Giorgio ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Native music on hold not playing on incoming calls
Giorgio Incantalupo schrieb: Hi, I'm trying to make native music on hold work on my Asterisk 1.2.9.1 server with a Sangoma PRI card. If I use a IAX phone connected to the PBX, I hear the music, but if I make a call from outside I hear nothing even if Asterisk console says music has started... it seems something related to zapata.conf but I cannot understand what's wrong. I also put musiconhold=native for every channel inside zapata.conf without success. Is there anybody who can help me, please? TIA Giorgio ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Did you Answer() the channel before playing the music? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Native music on hold not playing on incoming calls
So the configuration is under [native] in your musiconhold.conf? That does not mean it will use native MOH it is just a name, and you need to configur it properly See voip-info.org for configuration instructions. On 1/11/07, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi, I'm trying to make native music on hold work on my Asterisk 1.2.9.1 server with a Sangoma PRI card. If I use a IAX phone connected to the PBX, I hear the music, but if I make a call from outside I hear nothing even if Asterisk console says music has started... it seems something related to zapata.conf but I cannot understand what's wrong. I also put musiconhold=native for every channel inside zapata.conf without success. Is there anybody who can help me, please? TIA Giorgio ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] native music on hold distortion between files
I have native music on hold setup to play ulaw encoded files. No transcoding, caller is on a g.711u SIP channel. There is horrible distortion and noise between files for 1 to 2 seconds. Has anyone seen this? I check the files and trimmed silence from the end, the source of the noise is not the file. 1.2.13 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No music on hold?
I installed the asterisk-addons from source and installed them. It looks like it copied format_mp3.so but I'm not sure if 1.2.14 addons are compatible with asterisk 1.2.10. Also I unpacked the asterisk source for the 3 MOH .mp3's and copied them to the appropriate location. Still MOH is not working. Any other ideas? Thanks again, Phil Heya, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Do you have asterisk-addons installed? That could be the issue. J. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No music on hold?
Phil, did you add letter 'm' to your dial options?? exten = _XXX,1,Dial(SIP/XXX,60,m) Regards Arlen Nascimento On 12/20/06, Phil Finkler [EMAIL PROTECTED] wrote: I installed the asterisk-addons from source and installed them. It looks like it copied format_mp3.so but I'm not sure if 1.2.14 addons are compatible with asterisk 1.2.10. Also I unpacked the asterisk source for the 3 MOH .mp3's and copied them to the appropriate location. Still MOH is not working. Any other ideas? Thanks again, Phil Heya, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Do you have asterisk-addons installed? That could be the issue. J. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arlen Nascimento ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No music on hold?
On Tue, Dec 19, 2006 at 10:04:23PM -0500, Jerry wrote: Heya, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Do you have asterisk-addons installed? That could be the issue. Why? The OP is looking to play MP3s, and unless I misunderstood the instructions on the Wiki, addons is required (format_mp3) to play MP3's on 1.2.x. Is that not the case? J. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] No music on hold?
I already posted about this, but contrary to what is stated on the Wiki, mpg123 is required (at least in 1.2.x) if you wish to use mp3's for your MoH. I decided to go this route: http://www.voip-info.org/wiki/index.php?page=Asterisk+mpg123+faking+it -- Kevin Trumbull -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Sent: Wednesday, December 20, 2006 8:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No music on hold? On Tue, Dec 19, 2006 at 10:04:23PM -0500, Jerry wrote: Heya, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Do you have asterisk-addons installed? That could be the issue. Why? The OP is looking to play MP3s, and unless I misunderstood the instructions on the Wiki, addons is required (format_mp3) to play MP3's on 1.2.x. Is that not the case? J. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No music on hold?
On Wed, Dec 20, 2006 at 08:30:27AM -0800, Kevin Trumbull wrote: I already posted about this, but contrary to what is stated on the Wiki, mpg123 is required (at least in 1.2.x) if you wish to use mp3's for your MoH. I decided to go this route: http://www.voip-info.org/wiki/index.php?page=Asterisk+mpg123+faking+it If you're not streaming the MP# from an external source, converting it off-line will always be cheaper. And it may even actually save you disk space, because mp3 files have a much higher quality than Asterisk requires. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No music on hold?
No, I didn't have m added. Should I have it added? I know I've ran Asterisk with mp3123 in the past and music worked ok. It seems when I hit the hold button on the phones, it does trigger the message saying music on hold is starting but it INSTANTLY stops. I wish it gave some details as to WHY it stops suddenly! This is driving me nuts. Phil Phil, did you add letter 'm' to your dial options?? exten = _XXX,1,Dial(SIP/XXX,60,m) Regards Arlen Nascimento On 12/20/06, Phil Finkler PhilF at iqconsultinginc.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: I installed the asterisk-addons from source and installed them. It looks like it copied format_mp3.so but I'm not sure if 1.2.14 addons are compatible with asterisk 1.2.10. Also I unpacked the asterisk source for the 3 MOH .mp3's and copied them to the appropriate location. Still MOH is not working. Any other ideas? Thanks again, Phil Heya, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Do you have asterisk-addons installed? That could be the issue. J. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arlen Nascimento ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No music on hold?
Phil Finkler wrote: No, I didn’t have m added. Should I have it added? I know I’ve ran Asterisk with mp3123 in the past and music worked ok. It seems when I hit the hold button on the phones, it does trigger the message saying music on hold is starting but it INSTANTLY stops. I wish it gave some details as to WHY it stops suddenly! This is driving me nuts. Phil I was wondering the same thing as my MOH isn't working either in a 1.2.14 installation so I'm recompiling mpg123 as per: http://www.voip-info.org/tiki-index.php?page=Asterisk+mpg123+redhat We know you obviously need to use the m flag for the caller to hear MOH when dialing an extension, but I wonder if it's required AFTER the call has been answered and then put on hold. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No music on hold?
Lee Jenkins wrote: I was wondering the same thing as my MOH isn't working either in a 1.2.14 installation so I'm recompiling mpg123 as per: http://www.voip-info.org/tiki-index.php?page=Asterisk+mpg123+redhat We know you obviously need to use the m flag for the caller to hear MOH when dialing an extension, but I wonder if it's required AFTER the call has been answered and then put on hold. OK, asterisk just finished compiling and my MOH is working correctly. I have also verified that you do *not* have to have m in the Dial command in order for MOH to play when placed on hold. Note that I have a command in the initial context of my dialplan that set music on hold: exten=s,1,SetMusicOnHold(default) -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No music on hold?
I'm totally at a loss here. I can't get music on hold when placing someone on hold or when dialing an internal extension. When I dial an internal extension I hear ringing yet on my phone it shows little musical notes like it thinks it's hearing music. What to do! :-) Phil Lee Jenkins wrote: I was wondering the same thing as my MOH isn't working either in a 1.2.14 installation so I'm recompiling mpg123 as per: http://www.voip-info.org/tiki-index.php?page=Asterisk+mpg123+redhat We know you obviously need to use the m flag for the caller to hear MOH when dialing an extension, but I wonder if it's required AFTER the call has been answered and then put on hold. OK, asterisk just finished compiling and my MOH is working correctly. I have also verified that you do *not* have to have m in the Dial command in order for MOH to play when placed on hold. Note that I have a command in the initial context of my dialplan that set music on hold: exten=s,1,SetMusicOnHold(default) -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No music on hold?
Hi Phil, No, I didn't have m added. Should I have it added? I know I've ran Asterisk with mp3123 in the past and music worked ok. It seems when I hit the hold button on the phones, it does trigger the message saying music on hold is starting but it INSTANTLY stops. I wish it gave some details as to WHY it stops suddenly! This is driving me nuts. Do you see the mp3 format in your list of modules? (do a show modules from the CLI) I think this was what I had when the module wasn't loaded for me. The m option to dialing plays music instead of ringing while dialing, which (from my understanding) isn't what you were after. J. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No music on hold?
You should look at the asterisk-addons package. There is a addon module in the package called format_mp3 that will play your mp3 files instead of using mpg123 (which is a dead project). I just use sox to convert my mp3's to GSM with something like this: /usr/bin/sox musicfile.mp3 -r 8000 -c1 musicfile.gsm resample -ql This also puts it into 8bit mono, which sounds a little better on our phones. I have a compiled version of sox with added support for the lame encoder (mp3). They are RPM's built for CentOS/RHL. If you want them. On 12/20/06, Phil Finkler [EMAIL PROTECTED] wrote: I'm totally at a loss here. I can't get music on hold when placing someone on hold or when dialing an internal extension. When I dial an internal extension I hear ringing yet on my phone it shows little musical notes like it thinks it's hearing music. What to do! J Phil Lee Jenkins wrote: I was wondering the same thing as my MOH isn't working either in a 1.2.14 installation so I'm recompiling mpg123 as per: http://www.voip-info.org/tiki-index.php?page=Asterisk+mpg123+redhat We know you obviously need to use the m flag for the caller to hear MOH when dialing an extension, but I wonder if it's required AFTER the call has been answered and then put on hold. OK, asterisk just finished compiling and my MOH is working correctly. I have also verified that you do *not* have to have m in the Dial command in order for MOH to play when placed on hold. Note that I have a command in the initial context of my dialplan that set music on hold: exten=s,1,SetMusicOnHold(default) -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No music on hold?
On Wed, Dec 20, 2006 at 08:30:27AM -0800, Kevin Trumbull wrote: I already posted about this, but contrary to what is stated on the Wiki, mpg123 is required (at least in 1.2.x) if you wish to use mp3's for your MoH. I decided to go this route: http://www.voip-info.org/wiki/index.php?page=Asterisk+mpg123+faking+it Kevin - thanks for that link. We're now using native slin for music on hold, but it's a good resource. And thanks for pointing out the Wiki is actually wrong on this. :) If you're not streaming the MP# from an external source, converting it off-line will always be cheaper. And it may even actually save you disk space, because mp3 files have a much higher quality than Asterisk requires. Tzafir - both true, but I believe the OP was trying to play MP3's. It's kind of silly that Asterisk includes mp3 files, and then there isn't a way to play them. (There could be licence reasons for this, or another reason I'm not aware of, but people will assume that MP3's work.) And, there is a certain coolness factor in it just working, of course. Does format_mp3 only work properly in 1.4 then? I'll dig a bit and correct the wiki, if I can find a definitive answer. (I thought it was working on 1.2, but I've installed a few 1.4 boxes recently for testing, so my recollection could be of a 1.4 install). Thanks, J. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No music on hold?
Hi all, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Thanks, Phil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] No music on hold?
I had this same problem. I also read that mpg123 was not required, but it actually is if you wish to use mp3 files. I just decided to go with RAW files because I had problems converting some mp3's to the appropriate bit rate. http://www.voip-info.org/wiki/index.php?page=Asterisk+mpg123+faking+it -- Kevin Trumbull -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Finkler Sent: Tuesday, December 19, 2006 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No music on hold? Hi all, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Thanks, Phil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No music on hold?
On Tue, Dec 19, 2006 at 04:49:36PM -0500, Phil Finkler wrote: Hi all, I've got Asterisk 1.2.10 up and running on Debian using the back ports. Debian does not include the default MoH files that come with Debian for legal reasons. Get some sound files in the moh directory, basically, and use the naitve moh. Grab http://updates.xorcom.com/rapid/pool/main/f/freepbx/asterisk-sounds-moh-freepbx_2.1.3.dfsg-1.2902_all.deb or an equivalent. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No music on hold?
Heya, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Do you have asterisk-addons installed? That could be the issue. J. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No music on hold?
On Tue, Dec 19, 2006 at 10:04:23PM -0500, Jerry wrote: Heya, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Do you have asterisk-addons installed? That could be the issue. Why? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting Music On Hold working in * 1.2.12.1 with Fedora?
We are aware of the MPG123 tweaks that were always needed with Fedora in the past. We have MOH working on all other systems. We just installed a new system with a clean install of 1.2.12.1. It seems that there is info on the Wiki which states that there is a new way to do MOH using some internal Asterisk method. Says we have to install the add-ons package which we have done. I see no other hints or instructions on making MOH work with this version of Asterisk and Fedora 4. We only get silence where the MOH should be. Have I missed something? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting Music On Hold working in * 1.2.12.1 with Fedora?
voiplist wrote: We are aware of the MPG123 tweaks that were always needed with Fedora in the past. We have MOH working on all other systems. We just installed a new system with a clean install of 1.2.12.1. It seems that there is info on the Wiki which states that there is a new way to do MOH using some internal Asterisk method. Says we have to install the add-ons package which we have done. I see no other hints or instructions on making MOH work with this version of Asterisk and Fedora 4. We only get silence where the MOH should be. Have I missed something? Do you have timing? (ztdummy, zaptel, ..) ? Use native MOH instead. [default] mode=files directory=/var/lib/asterisk/moh-native random=yes ; Play the files in a random order in musiconhold.conf Convert your MOH files to the codec that you use. Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PROBLEM MUSIC ON HOLD
This is becasue the ztdummy drivers are not loaded. The procedure to load them is as follows: Switch to zaptel folder, i.e. type '/usr/src/zaptel-1.2.7' Edit the file named Makefile. If using vi, type 'vi Makefile' Search for ztdummy. To do so, type '/ztdummy' and press Enter key. It'll take you right there. You'll see that there is a '#' typed before 'ztdummy'. Press Insert on your keyboard, now you're in editing mode. Delete and '#' before 'ztdummy'. Press Escape key on your keyboard, then type ':qw" and press Enter key. The file is saved and you've left vi. Rebuild zaptel by tyuping 'make clean; make linux26; make install'. Also do 'make config' after installation is done. For more fine tuning, edit this zaptel file by typing 'vi /etc/sysconfig/zaptel'. Here are the zaptel modules listed. If you don't have any physical zaptel module, put a '#' before all the modules, except 'ztdummy' module. This step is optional, but recommended for saving some system resources. Reinstall Asterisk by switching to the /usr/src/asterisk directory and typing 'make clean;make; make install; make config' Reboot the computer by typing 'reboot'. If everything has gone smooth without any errors, MoH and other features which were not working before because of lack of a timer device, will start working. To check that ztdummy is loaded after reboot, give command 'lsmod' and see if ztdummy and zaptel are listed under the 'Modules' column, and ztdummy under the 'Used by' column in front of zaptel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PROBLEM MUSIC ON HOLD
I am tryin to make the asterisk work on my linux box but when I launch asterisk I get several warnings.I am using Fedora Core 3 (just installed and not yet updated with the latest) Asterisk Business Edition The error message I get is:Aug 7 18:24:42 WARNING[3416]: res_musiconhold.c:841 moh_register: Unable to open pseudo channel for timing... Sound may be choppy.Any idea what could be the problem? Elpidio Ramos PresidentRM International ServicesSA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax +52 (55)1755-6601 CellUSA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740Direct ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PROBLEM MUSIC ON HOLD
On Mon, Aug 07, 2006 at 04:26:28PM -0700, Elpidio Ramos wrote: I am tryin to make the asterisk work on my linux box but when I launch asterisk I get several warnings. I am using Fedora Core 3 (just installed and not yet updated with the latest) Asterisk Business Edition The error message I get is: Aug 7 18:24:42 WARNING[3416]: res_musiconhold.c:841 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. Any idea what could be the problem? I'd guess ztdummy hasn't been loaded. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?
I've noticed that native music on hold volume seems to be very loud sometimes. Is there anyway to turn this down? I know when using mpg123 I can set quietmp3 but what about when using native? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?
If you have to use it, make sure you only use the mpg123 bundled with the asterisk distribution. mpg123 from any other source (yes, evem the developer's website) will yield major issues. On 8-Jun-06, at 8:12 AM, Matt wrote: I've noticed that native music on hold volume seems to be very loud sometimes. Is there anyway to turn this down? I know when using mpg123 I can set quietmp3 but what about when using native? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?
I'm not using mpg123... I'm using NATIVE MOH! On 6/8/06, Jason Lixfeld [EMAIL PROTECTED] wrote: If you have to use it, make sure you only use the mpg123 bundled with the asterisk distribution. mpg123 from any other source (yes, evem the developer's website) will yield major issues. On 8-Jun-06, at 8:12 AM, Matt wrote: I've noticed that native music on hold volume seems to be very loud sometimes. Is there anyway to turn this down? I know when using mpg123 I can set quietmp3 but what about when using native? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?
- Matt [EMAIL PROTECTED] wrote: I'm not using mpg123... I'm using NATIVE MOH! No, the native file playback method does not offer any means to manipulate the volume of the sound being played. If you need to, you can edit the MOH files themselves using your tool of choice (sox, Audacity, etc.) to set the desired volume level. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?
Jason Lixfeld wrote: If you have to use it, make sure you only use the mpg123 bundled with the asterisk distribution. mpg123 from any other source (yes, evem the developer's website) will yield major issues. mpg123 is NOT bundled with Asteirsk. make mpg123 will DOWNLOAD the mpg123 source and compile it. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?
Kevin P. Fleming wrote: - Matt [EMAIL PROTECTED] wrote: I'm not using mpg123... I'm using NATIVE MOH! No, the native file playback method does not offer any means to manipulate the volume of the sound being played. If you need to, you can edit the MOH files themselves using your tool of choice (sox, Audacity, etc.) to set the desired volume level. With sox try -V 0.25 (or -v 0.25). I can't remember if it is an uppercase V or not. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?
K, That's what I thought... very odd.. all of our hold music (digium supplied or otherwise) seems to be very VERY loud. On 6/8/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: - Matt [EMAIL PROTECTED] wrote: I'm not using mpg123... I'm using NATIVE MOH! No, the native file playback method does not offer any means to manipulate the volume of the sound being played. If you need to, you can edit the MOH files themselves using your tool of choice (sox, Audacity, etc.) to set the desired volume level. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Many music on hold files
A feature we're often asked for in our ITSP product is to allow customers to upload their own music on hold, or to have it recorded for them by a recording studio with the latest news, weather, etc, punctuated by Welcome to customer, please hold. Since there may be thousands or tens of thousands of customers, and perhaps 10% of customers may want this feature with a couple of music on hold files each, there could easily be hundreds or even thousands of music on hold classes. Having all those mpg123 processes running is not really an option. To further complicate matters, there may be many Asterisk machines. The music on hold directory can be NFS mounted. However, we'd prefer not to have to update musiconhold.conf on each then tell each Asterisk to reload the file. Something database driven such as realtime would be nice. Can anyone please suggest a workable solution for this? Something like: SetMusicOnHoldFile(/path/to/file.wav) where the file was accessed only as needed, rather than an mpg123 process running all the time, would be ideal. This way our FastAGI daemon can look up the database at the start of the call, and set the correct file from an NFS mount. -- Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Many music on hold files
On 5/9/06, Alistair Cunningham [EMAIL PROTECTED] wrote: A feature we're often asked for in our ITSP product is to allow customers to upload their own music on hold, or to have it recorded for them by a recording studio with the latest news, weather, etc, punctuated by Welcome to customer, please hold. Since there may be thousands or tens of thousands of customers, and perhaps 10% of customers may want this feature with a couple of music on hold files each, there could easily be hundreds or even thousands of music on hold classes. Having all those mpg123 processes running is not really an option. To further complicate matters, there may be many Asterisk machines. The music on hold directory can be NFS mounted. However, we'd prefer not to have to update musiconhold.conf on each then tell each Asterisk to reload the file. Something database driven such as realtime would be nice. Can anyone please suggest a workable solution for this? Something like: SetMusicOnHoldFile(/path/to/file.wav) where the file was accessed only as needed, rather than an mpg123 process running all the time, would be ideal. This way our FastAGI daemon can look up the database at the start of the call, and set the correct file from an NFS mount. Use native music on hold mode. It will only use the resources then when it needs to. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Many music on hold files
BJ Weschke wrote: Use native music on hold mode. It will only use the resources then when it needs to. We considered this (though we didn't realise that it didn't use resources when not active, which is useful to know), but it would only really be practical if it could be configured from the database, preferably without needing to run Asterisk CLI or manager command when the database changes. We had a look at this, but both the Asterisk source and voip-info showed nothing useful. Do you know of a way to do this? Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Native music on hold on 1.0
Hi group! I have been using asterisk 1.2 for quite some time and now I need to go back on asterisk 1.0 (because of oh323 channel driver). One thing that I can't remember is does asterisk 1.0 support native music on hold? If it does, how can achieve it, because it doesn't work the same way as it does on asterisk 1.2. Thank you for your help. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native music on hold on 1.0
On Tue, 2006-04-11 at 09:26, Tomislav Parčina wrote: Hi group! I have been using asterisk 1.2 for quite some time and now I need to go back on asterisk 1.0 (because of oh323 channel driver). One thing that I can't remember is does asterisk 1.0 support native music on hold? If it does, how can achieve it, because it doesn't work the same way as it does on asterisk 1.2. Thank you for your help. -- Tomislav Parcina tparcina#lama.hr No it does not support it nativly. It uses the mpg123 program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multicast Music on Hold
Nathan Alberti wrote: As i am using a central asterisk box with multiple stub sites I don't wish every call put on hold to be wasting WAN bandwidth, I am wondering if it is possible to create a multicast stream to each site and rather than asterisk sending its address and the media information during a hold it sends the multicast address and multiple phones can be served by the one stream ? This is certainly possible, but I doubt that most existing SIP phones have the ability to subscribe to multicast groups and handle it properly. Without that, it would only work on a LAN (non-routed). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multicast Music on Hold
As I understand there is provisions for hold in both RFC 2543 (SIP) and RFC 3264 (SDP) and asterisk users the latter. The RFC 2543 method tells the UA its media stream is at 0.0.0.0 where as the method via SDP can tell it to listen to any address/port/ protocol combination, which is how asterisk tell it to listen to the audio stream it presents when it is asked to hold a call. Please stop me here if I have totally misunderstood the concept :) As i am using a central asterisk box with multiple stub sites I don't wish every call put on hold to be wasting WAN bandwidth, I am wondering if it is possible to create a multicast stream to each site and rather than asterisk sending its address and the media information during a hold it sends the multicast address and multiple phones can be served by the one stream ? Regards, Nathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Seperate music on hold for SIP extensions
I have a requirement to play different hold messages depending upon the extension that originated the call. I noticed a musicclass setting in sip.conf, but it appears this is global. I tried setting this on all of my individual extensions, but it didn't have any affect. Is there a way to achieve this, either through sip.conf or in the dial plan? Thanks, James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Native music on hold - Error
I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial(SIP/341-5931, SIP/344|20|wWtT) in new stack -- Called 344 -- SIP/344-5e4e is ringing -- SIP/344-5e4e answered SIP/341-5931 -- Attempting native bridge of SIP/341-5931 and SIP/344-5e4e -- Started music on hold, class 'native', on SIP/344-5e4e Mar 2 11:17:50 WARNING[7717]: format_wav.c:161 check_header: ot in mono 2 ar 2 11:17:50 WARNING[7717]: file.c:432 ast_filehelper: nable to open file on / var/lib/asterisk/moh-native/fpm-sunshine.wav ar 2 11:17:50 WARNING[7717]: res_musiconhold.c:225 ast_moh_files_next: nable to open file '/var/lib/asterisk/moh-native/fpm-sunshine': No such file or director y -- Stopped music on hold on SIP/344-5e4e == Spawn extension (sip, 344, 1) exited non-zero on 'SIP/341-5931' What have I done wrong? That file IS in that directory. When this starts to work I'll put more files in gsm and g729 format, but till then asterisk should encode this files. For this call I have use alaw codec. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native music on hold - Error
On 03/02/06 19:30 Tomislav Parèina said the following: What have I done wrong? That file IS in that directory. what are the file permissions/ownership and are they readable by the asterisk process ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Streaming Music On Hold
I spent a days or two on this and in the end did Musiconhold.conf [livestream1] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 -@ /etc/asterisk/stream.playlist Then in stream.playlist I just put the links from Shoutcast I wanted to use http://64.236.34.67:80/stream/1040 http://64.236.34.196:80/stream/1040 Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: 22 February 2006 21:18 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Streaming Music On Hold Thanks. I got it working. Yay. Now, it seems that Asterisk is very fussy with the streams. A lot don't work, especially when the URL ends in something.pls. Anyone know if that's true? Is Asterisk's support of this still pretty limited? Doug. -Original Message- From: Jonathan Augenstine [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 22, 2006 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Streaming Music On Hold Try this: musiconhold.conf: [stream2] mode=mp3 directory=http://pubint.ic.llnwd.net/stream/pubint_wnpr extensions.conf: exten = 1234,1,Answer exten = 1234,2,MusicOnHold(stream2) exten = 1234,3,Hangup On Wed, 2006-02-22 at 09:28 -0700, Douglas Garstang wrote: Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty application=http://pubint.ic.llnwd.net/stream/pubint_wnpr and this is how I am testing it: exten = 1234,1,Answer exten = 1234,2,SetMusiconHold(stream2) exten = 1234,3,WaitmusiconHold(60) exten = 1234,4,Hangup and this is the console output I get when I dial 1234: Asterisk Ready. *CLI -- Executing Answer(SIP/3250072-ed28, ) in new stack -- Executing SetMusicOnHold(SIP/3250072-ed28, stream2) in new stack -- Executing WaitMusicOnHold(SIP/3250072-ed28, 60) in new stack -- Started music on hold, class 'stream2', on channel 'SIP/3250072-ed28' -- Stopped music on hold on SIP/3250072-ed28 If I replace SetMusiconHold(stream2) with SetMusiconHold(default), I get the default music on hold. Running ngrep on port 80 shows me that the Asterisk system is not sending or receiving ANY data on port 80. What am I doing wrong? Yes, it has network and DNS connectivity. Can't believe it's this hard! Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Streaming Music On Hold
On Thu, 2006-02-23 at 08:09 +, Lee Archer wrote: I spent a days or two on this and in the end did Musiconhold.conf [livestream1] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 -@ /etc/asterisk/stream.playlist Then in stream.playlist I just put the links from Shoutcast I wanted to use That is useful to know, however that isnt the 'pls' play list files, they contain extra info over and above the url. however if one were to do something as simple as: grep ^File *.pls | cut -d= -f2- stream.playlist then they could use your information with pls files :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
