Re: [asterisk-users] Music on hold
Kris Stark wrote: Unfortunately, I was never asked about this to enough detail to be able to tell them how to set up the music, and as a result I have an eight minute file with several different messages all tied together into that one file. You can use Audacity to break that file into multiples http://audacity.sourceforge.net/ 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
Just split the file into multiple files n have it all uploaded to the same music on hold class. Now every time a caller is put on hold they will hear the files randomly. On 06-Mar-2015 8:32 AM, Kris Stark kris.st...@godataflow.com wrote: OK - so somebody just handed me the new music on hold file to use for the organization... Unfortunately, I was never asked about this to enough detail to be able to tell them how to set up the music, and as a result I have an eight minute file with several different messages all tied together into that one file. In general, we don't ever see a user being placed on hold for more than a minute, so using this file directly is of no use in general if I were to place it directly in to the server, as all users will only hear the first little bit of it. I suspect that when this was created, the producer assumed that the file would play in a loop, starting and stopping as callers were on hold. I realize that the streaming category will do just that, but since this is a local file, the setup works differently. (This is replacing a set of about 10 previous files that worked perfectly.) Is there any way, other than splitting up the file and trying to make decent segues between the files, to get this to work on a current version? I realize that getting it redone would be the best way, but I don't know if that is going to be an easy possibility. Any recommendations? Thanks! Kris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold
Hi
yes i have noticed the same result when i play a file like the default i can
hear the music but when i play another file there is no sound
about your question danny :yes i have created a file in
/var/lib/asterisk/moh1
and i configure in musiconhold.conf like below
[default1]
mode=files
directory=/var/lib/asterisk/moh1
2011/10/4 Kevin Oravits korav...@rcolegal.com
I’ve noticed on our system the sound files have to be in an exact format
for Asterisk to play them.
Bit Rate: 128kbps
Audio sample size: 16 bit
Channels: 1(mono)
Audio Sample rate: 8kHz
Audio format: PCM
** **
I actually downloaded a program and remixed the audio files to match these
settings. Before that, I couldn’t get my Asterisk to play any non-standard
music.
** **
*Kevin Oravits *
** **
*From:* Danny Nicholas [mailto:da...@debsinc.com]
*Sent:* Tuesday, October 04, 2011 11:48 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] music on hold
** **
You have files in /var/lib/asterisk/moh1?
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
elharit
*Sent:* Tuesday, October 04, 2011 12:49 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] music on hold
** **
i configure new music on hold like below in order to play music for outbond
calls
i want tp play a music until answer form customer
[default1]
mode=files
directory=/var/lib/asterisk/moh1
exten = 0678XX,1,Set(CALLERID(number)=520XX)
exten = 0678XX,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 0678XX,n,Dial(Zap/g1/${EXTEN},,m(default1))
exten = 0678XX,n,Hangup()
when i put the default music i can listen without issue but when i put
another music .wav Or gsm or Mp3
there is no music there is just the ringing
--
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Re: [asterisk-users] music on hold
Give that moh1 directory permissions, I once had similar issue that same
files being placed in default moh directory were played but making a new
call and directory couldn't play anything. So I fixed that by granting
directory permissions.
On Wed, Oct 5, 2011 at 2:25 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
Hi
yes i have noticed the same result when i play a file like the default i
can hear the music but when i play another file there is no sound
about your question danny :yes i have created a file in
/var/lib/asterisk/moh1
and i configure in musiconhold.conf like below
[default1]
mode=files
directory=/var/lib/asterisk/moh1
2011/10/4 Kevin Oravits korav...@rcolegal.com
I’ve noticed on our system the sound files have to be in an exact format
for Asterisk to play them.
Bit Rate: 128kbps
Audio sample size: 16 bit
Channels: 1(mono)
Audio Sample rate: 8kHz
Audio format: PCM
** **
I actually downloaded a program and remixed the audio files to match these
settings. Before that, I couldn’t get my Asterisk to play any non-standard
music.
** **
*Kevin Oravits *
** **
*From:* Danny Nicholas [mailto:da...@debsinc.com]
*Sent:* Tuesday, October 04, 2011 11:48 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] music on hold
** **
You have files in /var/lib/asterisk/moh1?
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
elharit
*Sent:* Tuesday, October 04, 2011 12:49 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] music on hold
** **
i configure new music on hold like below in order to play music for
outbond calls
i want tp play a music until answer form customer
[default1]
mode=files
directory=/var/lib/asterisk/moh1
exten = 0678XX,1,Set(CALLERID(number)=520XX)
exten =
0678XX,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 0678XX,n,Dial(Zap/g1/${EXTEN},,m(default1))
exten = 0678XX,n,Hangup()
when i put the default music i can listen without issue but when i put
another music .wav Or gsm or Mp3
there is no music there is just the ringing
--
_
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Re: [asterisk-users] music on hold
thanks for your replay i give the permissions 777 to this file moh1 and i
still have the same issue
best regards
2011/10/5 Sammy Govind govoi...@gmail.com
Give that moh1 directory permissions, I once had similar issue that same
files being placed in default moh directory were played but making a new
call and directory couldn't play anything. So I fixed that by granting
directory permissions.
On Wed, Oct 5, 2011 at 2:25 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
Hi
yes i have noticed the same result when i play a file like the default i
can hear the music but when i play another file there is no sound
about your question danny :yes i have created a file in
/var/lib/asterisk/moh1
and i configure in musiconhold.conf like below
[default1]
mode=files
directory=/var/lib/asterisk/moh1
2011/10/4 Kevin Oravits korav...@rcolegal.com
I’ve noticed on our system the sound files have to be in an exact
format for Asterisk to play them.
Bit Rate: 128kbps
Audio sample size: 16 bit
Channels: 1(mono)
Audio Sample rate: 8kHz
Audio format: PCM
** **
I actually downloaded a program and remixed the audio files to match
these settings. Before that, I couldn’t get my Asterisk to play any
non-standard music.
** **
*Kevin Oravits *
** **
*From:* Danny Nicholas [mailto:da...@debsinc.com]
*Sent:* Tuesday, October 04, 2011 11:48 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] music on hold
** **
You have files in /var/lib/asterisk/moh1?
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
elharit
*Sent:* Tuesday, October 04, 2011 12:49 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] music on hold
** **
i configure new music on hold like below in order to play music for
outbond calls
i want tp play a music until answer form customer
[default1]
mode=files
directory=/var/lib/asterisk/moh1
exten = 0678XX,1,Set(CALLERID(number)=520XX)
exten =
0678XX,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 0678XX,n,Dial(Zap/g1/${EXTEN},,m(default1))
exten = 0678XX,n,Hangup()
when i put the default music i can listen without issue but when i put
another music .wav Or gsm or Mp3
there is no music there is just the ringing
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] music on hold
You have files in /var/lib/asterisk/moh1?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Tuesday, October 04, 2011 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] music on hold
i configure new music on hold like below in order to play music for outbond
calls
i want tp play a music until answer form customer
[default1]
mode=files
directory=/var/lib/asterisk/moh1
exten = 0678XX,1,Set(CALLERID(number)=520XX)
exten = 0678XX,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 0678XX,n,Dial(Zap/g1/${EXTEN},,m(default1))
exten = 0678XX,n,Hangup()
when i put the default music i can listen without issue but when i put
another music .wav Or gsm or Mp3
there is no music there is just the ringing
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold
I've noticed on our system the sound files have to be in an exact format for
Asterisk to play them.
Bit Rate: 128kbps
Audio sample size: 16 bit
Channels: 1(mono)
Audio Sample rate: 8kHz
Audio format: PCM
I actually downloaded a program and remixed the audio files to match these
settings. Before that, I couldn't get my Asterisk to play any non-standard
music.
Kevin Oravits
From: Danny Nicholas [mailto:da...@debsinc.com]
Sent: Tuesday, October 04, 2011 11:48 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] music on hold
You have files in /var/lib/asterisk/moh1?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Tuesday, October 04, 2011 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] music on hold
i configure new music on hold like below in order to play music for outbond
calls
i want tp play a music until answer form customer
[default1]
mode=files
directory=/var/lib/asterisk/moh1
exten = 0678XX,1,Set(CALLERID(number)=520XX)
exten = 0678XX,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 0678XX,n,Dial(Zap/g1/${EXTEN},,m(default1))
exten = 0678XX,n,Hangup()
when i put the default music i can listen without issue but when i put another
music .wav Or gsm or Mp3
there is no music there is just the ringing
--
_
-- 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:
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Re: [asterisk-users] Music on Hold not working?
James Miller wrote: From the command you suggested to enter: Class: default File: /var/lib/asterisk/moh//reno_project-system You have 1 too many // in the directory structure. It should be: /var/lib/asterisk/moh/reno_project-system Not /var/lib/asterisk/moh//reno_project-system 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold not working?
On Sun, Jan 16, 2011 at 09:29:09AM -0500, Doug Lytle wrote: James Miller wrote: From the command you suggested to enter: Class: default File: /var/lib/asterisk/moh//reno_project-system You have 1 too many // in the directory structure. It should be: /var/lib/asterisk/moh/reno_project-system Not /var/lib/asterisk/moh//reno_project-system Is that really an issue? open() and all others would normally just reduce '//' to a '/'. -- 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] Music on Hold not working?
Tzafrir Cohen wrote: Is that really an issue? open() and all others would normally just reduce '//' to a '/'. That, I really wouldn't know. I'm not a programmer. I noted the differences between mine and his. 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold not working?
Well... Looks like he's trying to use a streaming MOH solution like an online radio station or something, so the files are irrelevant. Too bad the original post didn't specify that. I still think there is a different source selected for the call queue than for the rest of the system. Sorry for the top post... Blackberry won't do it any other way. Sent from my BlackBerry® smartphone -Original Message- From: Doug Lytle supp...@drdos.info Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 16 Jan 2011 10:50:48 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold not working? Tzafrir Cohen wrote: Is that really an issue? open() and all others would normally just reduce '//' to a '/'. That, I really wouldn't know. I'm not a programmer. I noted the differences between mine and his. 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold not working?
On Sat, Jan 15, 2011 at 09:20:10AM -0500, James Miller wrote: I'm going out on a limb here, as I'm still pretty new to Astrisk and running my own VOIP server, however I believe there is a bug or flaw with the Music on Hold feature. I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. Well, why should it play something different? How have you configured it to play something different? -- 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] Music on Hold not working?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, 16 January, 2011 12:47 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold not working? On Sat, Jan 15, 2011 at 09:20:10AM -0500, James Miller wrote: I'm going out on a limb here, as I'm still pretty new to Astrisk and running my own VOIP server, however I believe there is a bug or flaw with the Music on Hold feature. I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. Well, why should it play something different? How have you configured it to play something different? -- 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 [James M Miller] Well one would think that if you configure the Music on Hold feature by setting streams for it to pull from, it should play it no matter how the phone is dialed. Meaning if I dial another extension on the network, I should hear the MOH since I have it programmed with streams. However what is occurring is it is only playing when you are placed into a queue. Once someone picks up the line, it starts playing the default again if that person places the person on hold. One would think that it would play MOH no matter what if you have the streams programmed and override the defaults, at least that's what I'd like for it to do. Regards, James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold not working?
MOH plays the default class unless specified by a channel variable to play a different one. In queues.conf you can specify the MOH class on a queue by queue basis, but that's the hold music for someone waiting to be answered. Once an agent answers, if they put someone on hold they'll be put into the default MOH class unless a channel variable is specified beforehand. Thanks, --Warren Selby, dCAP On Jan 16, 2011, at 11:55 AM, James M Miller paramedi...@gmail.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, 16 January, 2011 12:47 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold not working? On Sat, Jan 15, 2011 at 09:20:10AM -0500, James Miller wrote: I'm going out on a limb here, as I'm still pretty new to Astrisk and running my own VOIP server, however I believe there is a bug or flaw with the Music on Hold feature. I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. Well, why should it play something different? How have you configured it to play something different? -- 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 [James M Miller] Well one would think that if you configure the Music on Hold feature by setting streams for it to pull from, it should play it no matter how the phone is dialed. Meaning if I dial another extension on the network, I should hear the MOH since I have it programmed with streams. However what is occurring is it is only playing when you are placed into a queue. Once someone picks up the line, it starts playing the default again if that person places the person on hold. One would think that it would play MOH no matter what if you have the streams programmed and override the defaults, at least that's what I'd like for it to do. Regards, James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold not working?
That's what I'm wanting to change. I want it to stream 100% of the time no matter if the person is in queue or if an agent has answered. Sent from my Verizon BlackBerry. Always on, Always Connected -Original Message- From: Warren Selby wcse...@selbytech.com Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 16 Jan 2011 12:41:31 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold not working? MOH plays the default class unless specified by a channel variable to play a different one. In queues.conf you can specify the MOH class on a queue by queue basis, but that's the hold music for someone waiting to be answered. Once an agent answers, if they put someone on hold they'll be put into the default MOH class unless a channel variable is specified beforehand. Thanks, --Warren Selby, dCAP On Jan 16, 2011, at 11:55 AM, James M Miller paramedi...@gmail.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, 16 January, 2011 12:47 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold not working? On Sat, Jan 15, 2011 at 09:20:10AM -0500, James Miller wrote: I'm going out on a limb here, as I'm still pretty new to Astrisk and running my own VOIP server, however I believe there is a bug or flaw with the Music on Hold feature. I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. Well, why should it play something different? How have you configured it to play something different? -- 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 [James M Miller] Well one would think that if you configure the Music on Hold feature by setting streams for it to pull from, it should play it no matter how the phone is dialed. Meaning if I dial another extension on the network, I should hear the MOH since I have it programmed with streams. However what is occurring is it is only playing when you are placed into a queue. Once someone picks up the line, it starts playing the default again if that person places the person on hold. One would think that it would play MOH no matter what if you have the streams programmed and override the defaults, at least that's what I'd like for it to do. Regards, James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold not working?
On Sat, Jan 15, 2011 at 7:20 AM, James Miller paramedi...@gmail.com wrote: I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. the middle, and still can not get MOH to work. Did you create /var/lib/asterisk/mohmp3/stream/stream.mp3? Did you Google it and try the solution here: http://nerdvittles.com/index.php?p=92 ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold not working?
On Sun, Jan 16, 2011 at 11:41 AM, Warren Selby wcse...@selbytech.com wrote: MOH plays the default class unless specified by a channel variable to play a different one. In queues.conf you can specify the MOH class on a queue by queue basis, but that's the hold music for someone waiting to be answered. Once an agent answers, if they put someone on hold they'll be put into the default MOH class unless a channel variable is specified beforehand. Thanks, --Warren Selby, dCAP On Jan 16, 2011, at 11:55 AM, James M Miller paramedi...@gmail.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, 16 January, 2011 12:47 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold not working? On Sat, Jan 15, 2011 at 09:20:10AM -0500, James Miller wrote: I'm going out on a limb here, as I'm still pretty new to Astrisk and running my own VOIP server, however I believe there is a bug or flaw with the Music on Hold feature. I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. Well, why should it play something different? How have you configured it to play something different? -- 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 [James M Miller] Well one would think that if you configure the Music on Hold feature by setting streams for it to pull from, it should play it no matter how the phone is dialed. Meaning if I dial another extension on the network, I should hear the MOH since I have it programmed with streams. However what is occurring is it is only playing when you are placed into a queue. Once someone picks up the line, it starts playing the default again if that person places the person on hold. One would think that it would play MOH no matter what if you have the streams programmed and override the defaults, at least that's what I'd like for it to do. Regards, James -- _ -- 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 Yep. musinonhold.conf has not had the default changed to streaming. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold not working?
Sorry for the top-post, on my phone... Then change the settings in the default class to what you want, or set a channel variable at the start of the call for the MusicOnHold class you want. Thanks, --Warren Selby, dCAP On Jan 16, 2011, at 12:44 PM, James Miller paramedi...@gmail.com wrote: That's what I'm wanting to change. I want it to stream 100% of the time no matter if the person is in queue or if an agent has answered. Sent from my Verizon BlackBerry. Always on, Always Connected -Original Message- From: Warren Selby wcse...@selbytech.com Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 16 Jan 2011 12:41:31 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold not working? MOH plays the default class unless specified by a channel variable to play a different one. In queues.conf you can specify the MOH class on a queue by queue basis, but that's the hold music for someone waiting to be answered. Once an agent answers, if they put someone on hold they'll be put into the default MOH class unless a channel variable is specified beforehand. Thanks, --Warren Selby, dCAP On Jan 16, 2011, at 11:55 AM, James M Miller paramedi...@gmail.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, 16 January, 2011 12:47 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold not working? On Sat, Jan 15, 2011 at 09:20:10AM -0500, James Miller wrote: I'm going out on a limb here, as I'm still pretty new to Astrisk and running my own VOIP server, however I believe there is a bug or flaw with the Music on Hold feature. I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. Well, why should it play something different? How have you configured it to play something different? -- 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 [James M Miller] Well one would think that if you configure the Music on Hold feature by setting streams for it to pull from, it should play it no matter how the phone is dialed. Meaning if I dial another extension on the network, I should hear the MOH since I have it programmed with streams. However what is occurring is it is only playing when you are placed into a queue. Once someone picks up the line, it starts playing the default again if that person places the person on hold. One would think that it would play MOH no matter what if you have the streams programmed and override the defaults, at least that's what I'd like for it to do. Regards, James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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
Re: [asterisk-users] Music on Hold not working?
I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. If it is playing the default music, then the MOH function is working. What do you get from moh show files in Asterisk? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold not working?
On Sat, Jan 15, 2011 at 9:36 AM, James Miller paramedi...@gmail.com wrote: Forgive me, but how do I do moh show files? Basically what is occurring is: If you enter a queue and are waiting to be answered, you will hear the streaming MOH If you call another extension on the system, you will only hear the default MOH. I want it to stream MOH for everything. Hopefully that makes sense. Regards, James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gary Allen *Sent:* Saturday, January 15, 2011 11:33 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Music on Hold not working? I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. If it is playing the default music, then the MOH function is working. What do you get from moh show files in Asterisk? Go into Asterisk CLI (asterisk -r) and issue the command moh show files. I don't see how you can have different MOH in a queue vs. being on hold unless you have specified a specific MOH group for your call queues. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold not working?
From the command you suggested to enter: Class: default File: /var/lib/asterisk/moh//reno_project-system File: /var/lib/asterisk/moh//macroform-robot_dity File: /var/lib/asterisk/moh//manolo_camp-morning_coffee File: /var/lib/asterisk/moh//macroform-cold_day File: /var/lib/asterisk/moh//macroform-the_simplicity Class: none File: /var/lib/asterisk/moh/.nomusic_reserved/silence Basically the queues will stream the online music, but if I call another extension on the network, it will play just the default sounds. One would think that if you have suggested the system play streaming music for everything else, it would follow suite and play streaming for ext to ext calls. I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Allen Sent: Saturday, January 15, 2011 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold not working? On Sat, Jan 15, 2011 at 9:36 AM, James Miller paramedi...@gmail.com wrote: Forgive me, but how do I do moh show files? Basically what is occurring is: If you enter a queue and are waiting to be answered, you will hear the streaming MOH If you call another extension on the system, you will only hear the default MOH. I want it to stream MOH for everything. Hopefully that makes sense. Regards, James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Allen Sent: Saturday, January 15, 2011 11:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold not working? I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. If it is playing the default music, then the MOH function is working. What do you get from moh show files in Asterisk? Go into Asterisk CLI (asterisk -r) and issue the command moh show files. I don't see how you can have different MOH in a queue vs. being on hold unless you have specified a specific MOH group for your call queues. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold Help
On my own version of sox (14.3.0), says -w option is not allowed ABEJIDE, Ayodele A. (CCNA) +2348039269311 Date: Mon, 1 Nov 2010 13:39:43 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music On Hold Help Un-top-posting... On Sun, 31 Oct 2010, Matt Darnell wrote: We have downloaded some royalty free music but it sounds 'fuzzy' when we test it with the system. On Sun, Oct 31, 2010 at 5:34 PM, Steve Edwards asterisk@sedwards.com wrote: Can you post a link to the original? On Sun, 31 Oct 2010, Matt Darnell wrote: Here is the original - http://www.makaicom.com/music/gt_30.wav Here is after we downsample using cool edit - http://www.makaicom.com/music/gt-30-ce.wav On Sun, Oct 31, 2010 at 9:43 PM, Steve Edwards Sounds reasonable to me. Do you have issues with all MOH? On Mon, 1 Nov 2010, Matt Darnell wrote: Did you use this syntax to convert: sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql Close, I don't do the resample effect. Way back when I wrote my transcribe script it didn't seem to add any goodness to my prompts. Once I have the file in the correct format, I use normalize (http://normalize.nongnu.org/) to adjust the volume. Lately, I've been receiving more prompts with visible (as seem in Audacity) DC offset so I'm looking for command line tools to correct that as well as high pass and low pass filters. I think the newer (than distributed with CentOS 5.x) version of sox will handle everything I need. Anybody care to volunteer their transcoding script? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold Help
On Sun, 31 Oct 2010, Matt Darnell wrote: We have downloaded some royalty free music but it sounds 'fuzzy' when we test it with the system. On Sun, Oct 31, 2010 at 5:34 PM, Steve Edwards asterisk@sedwards.com wrote: Can you post a link to the original? On Sun, 31 Oct 2010, Matt Darnell wrote: Here is the original - http://www.makaicom.com/music/gt_30.wav Here is after we downsample using cool edit - http://www.makaicom.com/music/gt-30-ce.wav Sounds reasonable to me. Do you have issues with all MOH? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold Help
Steve, Did you use this syntax to convert: sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql -Matt On Sun, Oct 31, 2010 at 9:43 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 31 Oct 2010, Matt Darnell wrote: We have downloaded some royalty free music but it sounds 'fuzzy' when we test it with the system. On Sun, Oct 31, 2010 at 5:34 PM, Steve Edwards asterisk@sedwards.com wrote: Can you post a link to the original? On Sun, 31 Oct 2010, Matt Darnell wrote: Here is the original - http://www.makaicom.com/music/gt_30.wav Here is after we downsample using cool edit - http://www.makaicom.com/music/gt-30-ce.wav Sounds reasonable to me. Do you have issues with all MOH? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold Help
Un-top-posting... On Sun, 31 Oct 2010, Matt Darnell wrote: We have downloaded some royalty free music but it sounds 'fuzzy' when we test it with the system. On Sun, Oct 31, 2010 at 5:34 PM, Steve Edwards asterisk@sedwards.com wrote: Can you post a link to the original? On Sun, 31 Oct 2010, Matt Darnell wrote: Here is the original - http://www.makaicom.com/music/gt_30.wav Here is after we downsample using cool edit - http://www.makaicom.com/music/gt-30-ce.wav On Sun, Oct 31, 2010 at 9:43 PM, Steve Edwards Sounds reasonable to me. Do you have issues with all MOH? On Mon, 1 Nov 2010, Matt Darnell wrote: Did you use this syntax to convert: sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql Close, I don't do the resample effect. Way back when I wrote my transcribe script it didn't seem to add any goodness to my prompts. Once I have the file in the correct format, I use normalize (http://normalize.nongnu.org/) to adjust the volume. Lately, I've been receiving more prompts with visible (as seem in Audacity) DC offset so I'm looking for command line tools to correct that as well as high pass and low pass filters. I think the newer (than distributed with CentOS 5.x) version of sox will handle everything I need. Anybody care to volunteer their transcoding script? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold Help
On Sun, 31 Oct 2010, Matt Darnell wrote: We have downloaded some royalty free music but it sounds 'fuzzy' when we test it with the system. Can you post a link to the original? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold Help
On Sun, Oct 31, 2010 at 5:34 PM, Steve Edwards asterisk@sedwards.comwrote: On Sun, 31 Oct 2010, Matt Darnell wrote: We have downloaded some royalty free music but it sounds 'fuzzy' when we test it with the system. Can you post a link to the original? Here is the original - http://www.makaicom.com/music/gt_30.wav Here is after we downsample using cool edit - http://www.makaicom.com/music/gt-30-ce.wav Appreciate any help. -Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold in blind transfer
On Fri, Aug 27, 2010 at 7:39 AM, Tino t...@sparksupport.com wrote: Is it possible to avoid playing music on hold during a blind transfer ? Please do not cross-post the same message to multiple lists. Yes, configure an empty MoH class or not loading MoH are some options, also: *CLI core show application Dial -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold 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! _ ¿Quieres descubrir todos los trucos de Windows 7? ¡Hazlo aquí! http://www.sietesunpueblodeexpertos.com/index_windows7.html-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold 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! http://serviciosmoviles.es.msn.com/hotmail/yoigo.aspx _ ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! http://www.ayudartepodria.com _ Del Lado Oscuro de Internet protegerte puedes. ¡Entra ya en www.ayudartepodria.com! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold problema
Hi, As Danny said, asterisk is looking for slin or ulaw files. Are your wav files in any of these formats? Did you just copied them from somewhere without changing their format? Also note they should be 8KHz mono 16 bit files. You can do this in a simple utility like Windows Recorder. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-18 10:35 AM, Danny Nicholas da...@debsinc.com wrote: 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 ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold 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! _ ¿Quieres descubrir todos los trucos de Windows 7? ¡Hazlo aquí! http://www.sietesunpueblodeexpertos.com/index_windows7.html-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold 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 http://serviciosmoviles.es.msn.com/hotmail/yoigo.aspx ya! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold 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! www.ayudartepodria.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold
Pressing hold on the telephone set may not be sending hold to Asterisk to trigger the correct action. You can verify this from the CLI. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur hasan Sent: Wednesday, May 26, 2010 1:34 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold Hello Yesterday, i brought linksys PAP2 and have success with that. The only thing that does not go well is the music on hold. When i press 'hold' button from the telephone set instead of playing the music on hold that i have setup in Asterisk, Telephone Set plays its own MOH. Is there any way to tackle this issue. Regards Taimur Hasan -THQ- !!!ONE _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign https://signup.live.com/signup.aspx?id=60969 up now. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold
Hello Ya you are right moh in Asterisk is not triggered. Is there any solution to that ? Regards Taimur Hasan -THQ- !!!ONE From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 26 May 2010 13:39:00 -0500 Subject: Re: [asterisk-users] Music on Hold Pressing “hold” on the telephone set may not be sending “hold” to Asterisk to trigger the correct action. You can verify this from the CLI. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur hasan Sent: Wednesday, May 26, 2010 1:34 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold Hello Yesterday, i brought linksys PAP2 and have success with that. The only thing that does not go well is the music on hold. When i press 'hold' button from the telephone set instead of playing the music on hold that i have setup in Asterisk, Telephone Set plays its own MOH. Is there any way to tackle this issue. Regards Taimur Hasan -THQ- !!!ONE Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold
Either teach your operators to press a new sequence that will send hold to asterisk or reprogram your phone. I know how to do with Polycom phones, but not linksys. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur hasan Sent: Wednesday, May 26, 2010 1:46 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold Hello Ya you are right moh in Asterisk is not triggered. Is there any solution to that ? Regards Taimur Hasan -THQ- !!!ONE _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 26 May 2010 13:39:00 -0500 Subject: Re: [asterisk-users] Music on Hold Pressing hold on the telephone set may not be sending hold to Asterisk to trigger the correct action. You can verify this from the CLI. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur hasan Sent: Wednesday, May 26, 2010 1:34 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold Hello Yesterday, i brought linksys PAP2 and have success with that. The only thing that does not go well is the music on hold. When i press 'hold' button from the telephone set instead of playing the music on hold that i have setup in Asterisk, Telephone Set plays its own MOH. Is there any way to tackle this issue. Regards Taimur Hasan -THQ- !!!ONE _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. https://signup.live.com/signup.aspx?id=60969 _ Hotmail: Free, trusted and rich email service. Get it now. https://signup.live.com/signup.aspx?id=60969 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold
You would need to see if there is a hook flash hold. Try playing with a hook/flash ( ie do a flash, wait, then hangup phone, it may send the onhold message ) it may also ring back. Or you will have to park the call Hook flash , Dial 700 (if that's your park extension), hangup, then recall the call if you want to answer it. An analog port is usefully on hook or off hook, no on hold unless the ATA has something documented for it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur hasan Sent: Wednesday, May 26, 2010 2:46 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold Hello Ya you are right moh in Asterisk is not triggered. Is there any solution to that ? Regards Taimur Hasan -THQ- !!!ONE _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 26 May 2010 13:39:00 -0500 Subject: Re: [asterisk-users] Music on Hold Pressing hold on the telephone set may not be sending hold to Asterisk to trigger the correct action. You can verify this from the CLI. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur hasan Sent: Wednesday, May 26, 2010 1:34 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold Hello Yesterday, i brought linksys PAP2 and have success with that. The only thing that does not go well is the music on hold. When i press 'hold' button from the telephone set instead of playing the music on hold that i have setup in Asterisk, Telephone Set plays its own MOH. Is there any way to tackle this issue. Regards Taimur Hasan -THQ- !!!ONE _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. https://signup.live.com/signup.aspx?id=60969 _ Hotmail: Free, trusted and rich email service. Get it now. https://signup.live.com/signup.aspx?id=60969 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold
hello, which phone do you have behind the pap2 cause the hook flash time sometimes could be set in the phone and then it will work with the pap2 also. you should have a look at spaconfig.de (its a german website) but the default parameters in sip and regional conf, may help you. best regards steve taimur hasan schrieb: Hello Yesterday, i brought linksys PAP2 and have success with that. The only thing that does not go well is the music on hold. When i press 'hold' button from the telephone set instead of playing the music on hold that i have setup in Asterisk, Telephone Set plays its own MOH. Is there any way to tackle this issue. Regards Taimur Hasan *-THQ- !!!ONE* Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. https://signup.live.com/signup.aspx?id=60969 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold
thx a lot friend. On Sat, 21 Nov 2009 20:08:45 -0500, C F shma...@gmail.com wrote: On Thu, Nov 19, 2009 at 10:31 PM, aster...@opensourcesolution.in wrote: hello friends i want very simple thing in my dial plan. 1.When ever calls come at exten 2000 and if it is not answered with in 60 secs it should hangup. Set absolute timeout to 60 seconds. 2.when ever call comes at exten 2000 and if it is answered within 60 secs and if person who receives the call, puts the call on hold than music on hold should begins. Setup music on hold: http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf 3.if music on hold is placed for more than 60 secs call should hangup. As far as I know, that is impossible to do with current code, since asterisk sees an answered call the same way a call thats place on hold, therefore asterisk has no way to distinguish between being on hold or actively talking on the phone. my extention.conf is like this vi /etc/asterisk/extentions.conf exten = 2000,1,Answer() exten = 2000,n,Dial(SIP/2000,60) exten = 2000,n,Dial(SIP/2000,60,m) exten = 2000,n,Hangup the output of this is that when call is coming at exten 2000 call is answered and another call comes n first call is on hold after 60 secs music on hold starts but if i receive call before 60 secs even than MOH starts even i dont put call on hold. thx ___ -- 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] music on hold
On Thu, Nov 19, 2009 at 10:31 PM, aster...@opensourcesolution.in wrote: hello friends i want very simple thing in my dial plan. 1.When ever calls come at exten 2000 and if it is not answered with in 60 secs it should hangup. Set absolute timeout to 60 seconds. 2.when ever call comes at exten 2000 and if it is answered within 60 secs and if person who receives the call, puts the call on hold than music on hold should begins. Setup music on hold: http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf 3.if music on hold is placed for more than 60 secs call should hangup. As far as I know, that is impossible to do with current code, since asterisk sees an answered call the same way a call thats place on hold, therefore asterisk has no way to distinguish between being on hold or actively talking on the phone. my extention.conf is like this vi /etc/asterisk/extentions.conf exten = 2000,1,Answer() exten = 2000,n,Dial(SIP/2000,60) exten = 2000,n,Dial(SIP/2000,60,m) exten = 2000,n,Hangup the output of this is that when call is coming at exten 2000 call is answered and another call comes n first call is on hold after 60 secs music on hold starts but if i receive call before 60 secs even than MOH starts even i dont put call on hold. thx ___ -- 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] music on hold
By default, if a Dial() is successful, further execution of additional priorities in the dial plan is aborted. It sounds like after the call completes, you're wanting execution continue to fall through. You need to use the 'g' flag to Dial, as explained here: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Example: Dial(SIP/2000,60,g) aster...@opensourcesolution.in wrote: hello friends i want very simple thing in my dial plan. 1.When ever calls come at exten 2000 and if it is not answered with in 60 secs it should hangup. 2.when ever call comes at exten 2000 and if it is answered within 60 secs and if person who receives the call, puts the call on hold than music on hold should begins. 3.if music on hold is placed for more than 60 secs call should hangup. my extention.conf is like this vi /etc/asterisk/extentions.conf exten = 2000,1,Answer() exten = 2000,n,Dial(SIP/2000,60) exten = 2000,n,Dial(SIP/2000,60,m) exten = 2000,n,Hangup the output of this is that when call is coming at exten 2000 call is answered and another call comes n first call is on hold after 60 secs music on hold starts but if i receive call before 60 secs even than MOH starts even i dont put call on hold. thx ___ -- 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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] music on hold
aster...@opensourcesolution.in wrote: i had done r/d of voice mail in which i got succes, now when i call exten 2000 and it on hold there is no music on hold. plz guide me what mistakes i am doing. Have you made the necessary adjustments to /etc/asterisk/musiconhold.conf to define music contexts, where the input files come from, etc? MoH doesn't get generated magically. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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] music on hold
Pawan, I am getting the sense that you do not understand how to use a mailing list, so I will take the time to uh, guide you before answering your question as best as I can. u r welcome The purpose of making a posting to a mailing list is not to find a single person that replies to you on the list and then bother them individually at their personal address. Instead, you should reply back to the list (the default Reply-To address) and continue the discussion there; this is known as creating a discussion thread. There are two main reasons for doing this: 1) Many people have their mail clients set up to deal with mailing list traffic in particular ways, and consider your personal inquiries spam; 2) When discussion is kept on the list as opposed to taken off of it, everyone who is subscribed benefits from the information exchanged, can add new contributions. More importantly, someone else new to Asterisk who is in your position later on - but perhaps slightly more resourceful than you are - can search the list archives on Google and benefit from this discussion. As far as your question, it seems to me that you did everything right. -- Alex aster...@opensourcesolution.in wrote: hi these r the steps to rn music on if i doing some mistake than plz guide me. *1-Install SOX (Sound Exchange Quality)* *yum install sox* * * *2 install a music (plz give the link to download)* * * *3 convert the song in asterisk format, convert it through sox* * * *4 file saved in mohwav* * * *5 now give the path in /etc/asterisk/musiconhold* thx -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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] music on hold
Peter Evans wrote: On Tue, Nov 11, 2008 at 02:35:19PM +0800, 邱磊 wrote: hii guys: i get the message from the asterisk: Started music on hold, class 'default', on Local/s...@skype-web-callback-dial-263to263-1775,1 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav [2008-11-11 14:32:41] WARNING[1781]: res_musiconhold.c:259 ast_moh_files_next: Unable to open file '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory This could be a tough one. If you can't solve this without help, should you really be playing with ancient scrolls of wisdom in the first place? Actually Peter, this isn't the real giveaway as to what the problem is. That would be the line that says [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 The wave files aren't properly encoded for Asterisk. ___ -- 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] Music On Hold
Hello all, I adressed this issue a couple of weekes ago, but didn't find a solution yet. It seems that MOH is not initiated by Asterisk version 1.6.1.6 when receiving sendonly INVITE, in order to put a call on hold. I configured features.conf with the following settings, and *9 initiates a proper HOLD with MOH, and so does *0 for blind transfer, but the a=sendonly doesn't: [featuremap] blindxfer = *0 atxfer = *9 Does anyone know if there's a parameter to set or if it's a bug in this version? Thanks. Original Message Subject: Re: [asterisk-users] Music On Hold From: Cyprus VoIP voi...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Saturday, 03 October, 2009 09:28:20 What does your musiconhold.conf look like? [general] [default] mode=files directory=/var/lib/asterisk/moh ___ -- 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] Music On Hold
What does your musiconhold.conf look like? [general] [default] mode=files directory=/var/lib/asterisk/moh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Music On Hold
Hi, I deleted all the default files and put one that I know that works on another Asterisk, but since then, I recompiled Asterisk and the default files were added. In order to test moh, I created a context for it: [default] exten = 888,1,Goto(moh,s,1) [moh] exten = s,1,Answer exten = s,2,MusicOnHold() When we dial 888, we hear the music and this appears in the console: -- Executing [...@default:1] Goto(SIP/24-08650e80, moh,s,1) in new stack -- Goto (moh,s,1) -- Executing [...@moh:1] Answer(SIP/24-08650e80, ) in new stack -- Executing [...@moh:2] MusicOnHold(SIP/24-08650e80, ) in new stack -- Started music on hold, class 'default', on SIP/24-08650e80 -- Stopped music on hold on SIP/24-08650e80 == Spawn extension (moh, s, 2) exited non-zero on 'SIP/24-08650e80' But, when I just put a call on hold, nothing is played and nothing appears in the console. I have no idea why this happens and what to do about it. Any suggestions? Thanks. Original Message Subject: Re: [asterisk-users] Music On Hold From: John A. Sullivan III jsulli...@opensourcedevel.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, 30 September, 2009 15:27:28 On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote: snip You see the wav files but do you see the files encoded for the codecs you are using? There's only one wav file there. No encoded files, but on asterisk 1.2 we have, it's the same file and it works. snip Hmm . . only one wav file. We had several. As I recall now, we actually installed 1.6.1.1 and upgraded. 1.6.1.1 had the old hold music. 1.6.1.6 has the new hold music. But I believe there are several files. Is that wav file valid, i.e., if you copy it to a system with a sound card and play it, does it play? Could it have been corrupted in copying or have incorrect permissions? - John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Music On Hold
Hello Cyprus, What is the output of moh files show CLI command ? Best regards, Ioan (Nini) Indreias www.modulo.ro On Fri, Oct 2, 2009 at 11:46 AM, Cyprus VoIP voi...@gmail.com wrote: Hi, I deleted all the default files and put one that I know that works on another Asterisk, but since then, I recompiled Asterisk and the default files were added. In order to test moh, I created a context for it: [default] exten = 888,1,Goto(moh,s,1) [moh] exten = s,1,Answer exten = s,2,MusicOnHold() When we dial 888, we hear the music and this appears in the console: -- Executing [...@default:1] Goto(SIP/24-08650e80, moh,s,1) in new stack -- Goto (moh,s,1) -- Executing [...@moh:1] Answer(SIP/24-08650e80, ) in new stack -- Executing [...@moh:2] MusicOnHold(SIP/24-08650e80, ) in new stack -- Started music on hold, class 'default', on SIP/24-08650e80 -- Stopped music on hold on SIP/24-08650e80 == Spawn extension (moh, s, 2) exited non-zero on 'SIP/24-08650e80' But, when I just put a call on hold, nothing is played and nothing appears in the console. I have no idea why this happens and what to do about it. Any suggestions? Thanks. Original Message Subject: Re: [asterisk-users] Music On Hold From: John A. Sullivan III jsulli...@opensourcedevel.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, 30 September, 2009 15:27:28 On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote: snip You see the wav files but do you see the files encoded for the codecs you are using? There's only one wav file there. No encoded files, but on asterisk 1.2 we have, it's the same file and it works. snip Hmm . . only one wav file. We had several. As I recall now, we actually installed 1.6.1.1 and upgraded. 1.6.1.1 had the old hold music. 1.6.1.6 has the new hold music. But I believe there are several files. Is that wav file valid, i.e., if you copy it to a system with a sound card and play it, does it play? Could it have been corrupted in copying or have incorrect permissions? - John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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
Re: [asterisk-users] Music On Hold
What is the output of moh files show CLI command ? pbx*CLI moh show files Class: default File: /var/lib/asterisk/moh/manolo_camp-morning_coffee File: /var/lib/asterisk/moh/macroform-the_simplicity File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/music_100 File: /var/lib/asterisk/moh/CHANGES-asterisk-moh-opsound-2 File: /var/lib/asterisk/moh/CREDITS-asterisk-moh-opsound-2 File: /var/lib/asterisk/moh/LICENSE-asterisk-moh-opsound-2 One more thing I tried is to add the m option in the dial command, to check what happens, when the call is initially originated: exten = ,n,Dial(SIP/21SIP/22SIP/23SIP/24SIP/25,,m(default)) This is what the console shows me: -- Executing [1...@default:6] Dial(SIP/IN-PROXY-08563908, SIP/21SIP/22SIP/23SIP/24SIP/25,,m(default)) in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 -- Called 21 == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 -- Called 22 == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 -- Called 23 == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 -- Called 24 == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 -- Called 25 -- Started music on hold, class 'default', on SIP/IN-PROXY-08563908 -- SIP/23-b7c2d928 is ringing -- SIP/24-b7a7b3a8 is ringing -- SIP/22-b7c35cd8 is ringing -- SIP/21-b7c3a508 is ringing -- SIP/25-b7a83350 is ringing -- Stopped music on hold on SIP/IN-PROXY-08563908 == Spawn extension (default, 99935709, 6) exited non-zero on 'SIP/IN-PROXY-08563908' The music is played instead of the RBT, but during the conversation, when put on hold, I only get silence and I don't get any reference in the console to the fact that the call has been put on hold. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Music On Hold
What does your musiconhold.conf look like? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP Sent: Friday, October 02, 2009 3:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music On Hold Hi, I deleted all the default files and put one that I know that works on another Asterisk, but since then, I recompiled Asterisk and the default files were added. In order to test moh, I created a context for it: [default] exten = 888,1,Goto(moh,s,1) [moh] exten = s,1,Answer exten = s,2,MusicOnHold() When we dial 888, we hear the music and this appears in the console: -- Executing [...@default:1] Goto(SIP/24-08650e80, moh,s,1) in new stack -- Goto (moh,s,1) -- Executing [...@moh:1] Answer(SIP/24-08650e80, ) in new stack -- Executing [...@moh:2] MusicOnHold(SIP/24-08650e80, ) in new stack -- Started music on hold, class 'default', on SIP/24-08650e80 -- Stopped music on hold on SIP/24-08650e80 == Spawn extension (moh, s, 2) exited non-zero on 'SIP/24-08650e80' But, when I just put a call on hold, nothing is played and nothing appears in the console. I have no idea why this happens and what to do about it. Any suggestions? Thanks. Original Message Subject: Re: [asterisk-users] Music On Hold From: John A. Sullivan III jsulli...@opensourcedevel.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, 30 September, 2009 15:27:28 On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote: snip You see the wav files but do you see the files encoded for the codecs you are using? There's only one wav file there. No encoded files, but on asterisk 1.2 we have, it's the same file and it works. snip Hmm . . only one wav file. We had several. As I recall now, we actually installed 1.6.1.1 and upgraded. 1.6.1.1 had the old hold music. 1.6.1.6 has the new hold music. But I believe there are several files. Is that wav file valid, i.e., if you copy it to a system with a sound card and play it, does it play? Could it have been corrupted in copying or have incorrect permissions? - John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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
Re: [asterisk-users] Music On Hold
Hello, We posted the question below yesterday, but got no answer from the community. When we checked the same behavior with Asterisk 1.2, we got the Started music on hold, class... message on the console, but in 1.6, we get absolutely nothing. I tried to unload and reload the moh module and everything seems normal, but Asterisk still doesn't respond in the console to the HOLD action, represented by the INVITE message. the call itself is being placed on hold and can be retrieved, but the audio file is not played and the held party hears only a silence. If anyone knows how to debug/fix it, your help would be HIGHLY appreciated. We're really stuck. Thank you all in advance. Original Message Subject: Music On Hold From: Cyprus VoIP voi...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, 29 September, 2009 14:31:28 Hello, We need help in debugging Music On Hold on our Asterisk 1.6.1.6 From the SIP debug, I see that an extension sends an INVITE of the call to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but I don't see in the console any reference to the call being placed on hold. When I typed moh show files, I see the wav files of the /var/lib/asterisk/moh folder. How can I debug this? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Music On Hold
I'm afraid I can't be much help as I am both a newbie and it works just fine for me on 1.6.1.6. Of course, mine was a fresh installation. Is there anything in the logs to give you a clue? You see the wav files but do you see the files encoded for the codecs you are using? I think Asterisk will transcode on the fly but I'm not sure. Sorry - John On Wed, 2009-09-30 at 11:52 +0300, Cyprus VoIP wrote: Hello, We posted the question below yesterday, but got no answer from the community. When we checked the same behavior with Asterisk 1.2, we got the Started music on hold, class... message on the console, but in 1.6, we get absolutely nothing. I tried to unload and reload the moh module and everything seems normal, but Asterisk still doesn't respond in the console to the HOLD action, represented by the INVITE message. the call itself is being placed on hold and can be retrieved, but the audio file is not played and the held party hears only a silence. If anyone knows how to debug/fix it, your help would be HIGHLY appreciated. We're really stuck. Thank you all in advance. Original Message Subject: Music On Hold From: Cyprus VoIP voi...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, 29 September, 2009 14:31:28 Hello, We need help in debugging Music On Hold on our Asterisk 1.6.1.6 From the SIP debug, I see that an extension sends an INVITE of the call to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but I don't see in the console any reference to the call being placed on hold. When I typed moh show files, I see the wav files of the /var/lib/asterisk/moh folder. How can I debug this? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Music On Hold
I'm afraid I can't be much help as I am both a newbie and it works just fine for me on 1.6.1.6. Of course, mine was a fresh installation. Thanks for your help, John. Mine is also a fresh installation, but now at least I know it's not a version issue. Is there anything in the logs to give you a clue? There's absolutely nothing in the logs, and that's what surprises me. You see the wav files but do you see the files encoded for the codecs you are using? There's only one wav file there. No encoded files, but on asterisk 1.2 we have, it's the same file and it works. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Music On Hold
On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote: snip You see the wav files but do you see the files encoded for the codecs you are using? There's only one wav file there. No encoded files, but on asterisk 1.2 we have, it's the same file and it works. snip Hmm . . only one wav file. We had several. As I recall now, we actually installed 1.6.1.1 and upgraded. 1.6.1.1 had the old hold music. 1.6.1.6 has the new hold music. But I believe there are several files. Is that wav file valid, i.e., if you copy it to a system with a sound card and play it, does it play? Could it have been corrupted in copying or have incorrect permissions? - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Music on Hold
Just a shot in the dark but could MOH be choking on the long file names? (does it work on fred_chopin_pol_1)? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Saul Sent: Wednesday, September 16, 2009 4:18 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI moh show files Class: default File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1 These files were generated by SoX: Channels : 1 Sample Rate: 8000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM Endian Type: little Reverse Nibbles: no Reverse Bits : no Comment: 'Processed by SoX' This prints in the asterisk console when you attempt to put someone in hold: -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rv): == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] Goto(SIP/ATA-xx-L1-024b6d88, 1xx,1) in new stack -- Goto (phones,1xx,1) -- Executing [1xxx...@phones:1] MSet(SIP/ATA-xx-L1-024b6d88, oldcidnum=0) in new stack -- Executing [1xxx...@phones:2] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(name)=) in new stack -- Executing [1xxx...@phones:3] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(num)=xx) in new stack -- Executing [1xxx...@phones:4] Monitor(SIP/ATA-xx-L1-024b6d88, wav,/tmp/out 0 2009-09-17 03h 04m 51s CST xx,m) in new stack -- Executing [1xxx...@phones:5] Gosub(SIP/ATA-xx-L1-024b6d88, ExternalDial,s,1(1xx)) in new stack -- Executing [...@externaldial:1] MSet(SIP/ATA-xx-L1-024b6d88, LOCAL(num)=1xx) in new stack -- Executing [...@externaldial:2] MSet(SIP/ATA-xx-L1-024b6d88, ~~EXTEN~~=s) in new stack -- Executing [...@externaldial:3] Dial(SIP/ATA-xx-L1-024b6d88, SIP/1xxx...@link2voip-sw1,120) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx...@link2voip-sw1 -- SIP/link2voip-sw1-02477668 is making progress passing it to SIP/ATA-xx-L1-024b6d88 -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88 -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 doing dnsmgr_lookup for 'sip.ca2.link2voip.com' doing dnsmgr_lookup for 'sip.ca1.link2voip.com' == Spawn extension (ExternalDial, s, 3) exited non-zero on 'SIP/ATA-xx-L1-024b6d88' Any thoughts or ideas? If there were an error I could work on solving that, but there is none... Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Music on Hold
That was a good shot in the dark, but sadly renaming it to something simple (and removing all non ascii in the process) does not correct this. On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas da...@debsinc.com wrote: Just a “shot in the dark” but could MOH be choking on the “long file names”? (does it work on fred_chopin_pol_1)? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul *Sent:* Wednesday, September 16, 2009 4:18 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Music on Hold Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI moh show files Class: default File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1 These files were generated by SoX: Channels : 1 Sample Rate: 8000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM Endian Type: little Reverse Nibbles: no Reverse Bits : no Comment: 'Processed by SoX' This prints in the asterisk console when you attempt to put someone in hold: -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rv): == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] Goto(SIP/ATA-xx-L1-024b6d88, 1xx,1) in new stack -- Goto (phones,1xx,1) -- Executing [1xxx...@phones:1] MSet(SIP/ATA-xx-L1-024b6d88, oldcidnum=0) in new stack -- Executing [1xxx...@phones:2] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(name)=) in new stack -- Executing [1xxx...@phones:3] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(num)=xx) in new stack -- Executing [1xxx...@phones:4] Monitor(SIP/ATA-xx-L1-024b6d88, wav,/tmp/out 0 2009-09-17 03h 04m 51s CST xx,m) in new stack -- Executing [1xxx...@phones:5] Gosub(SIP/ATA-xx-L1-024b6d88, ExternalDial,s,1(1xx)) in new stack -- Executing [...@externaldial:1] MSet(SIP/ATA-xx-L1-024b6d88, LOCAL(num)=1xx) in new stack -- Executing [...@externaldial:2] MSet(SIP/ATA-xx-L1-024b6d88, ~~EXTEN~~=s) in new stack -- Executing [...@externaldial:3] Dial(SIP/ATA-xx-L1-024b6d88, SIP/1xxx...@link2voip-sw1,120) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx...@link2voip-sw1 -- SIP/link2voip-sw1-02477668 is making progress passing it to SIP/ATA-xx-L1-024b6d88 -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88 -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 doing dnsmgr_lookup for 'sip.ca2.link2voip.com' doing dnsmgr_lookup for 'sip.ca1.link2voip.com' == Spawn extension (ExternalDial, s, 3) exited non-zero on 'SIP/ATA-xx-L1-024b6d88' Any thoughts or ideas? If there were an error I could work on solving that, but there is none... Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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
Re: [asterisk-users] Music on Hold
Dan Saul escribió: Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI moh show files Class: default File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1 I would use a more friendly filename. That special accents and spaces maybe are confusing asterisk when it tries to read the files. Try renaming to chopin_op40-1 and chopin_op40-2 for example. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Music on Hold
What are your actual file names (/etc/asterisk/musiconhold/Frederic Chopin Polonaised Op. 40-2.wav?) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Saul Sent: Wednesday, September 16, 2009 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold That was a good shot in the dark, but sadly renaming it to something simple (and removing all non ascii in the process) does not correct this. On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas da...@debsinc.com wrote: Just a shot in the dark but could MOH be choking on the long file names? (does it work on fred_chopin_pol_1)? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Saul Sent: Wednesday, September 16, 2009 4:18 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI moh show files Class: default File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1 These files were generated by SoX: Channels : 1 Sample Rate: 8000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM Endian Type: little Reverse Nibbles: no Reverse Bits : no Comment: 'Processed by SoX' This prints in the asterisk console when you attempt to put someone in hold: -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rv): == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] Goto(SIP/ATA-xx-L1-024b6d88, 1xx,1) in new stack -- Goto (phones,1xx,1) -- Executing [1xxx...@phones:1] MSet(SIP/ATA-xx-L1-024b6d88, oldcidnum=0) in new stack -- Executing [1xxx...@phones:2] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(name)=) in new stack -- Executing [1xxx...@phones:3] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(num)=xx) in new stack -- Executing [1xxx...@phones:4] Monitor(SIP/ATA-xx-L1-024b6d88, wav,/tmp/out 0 2009-09-17 03h 04m 51s CST xx,m) in new stack -- Executing [1xxx...@phones:5] Gosub(SIP/ATA-xx-L1-024b6d88, ExternalDial,s,1(1xx)) in new stack -- Executing [...@externaldial:1] MSet(SIP/ATA-xx-L1-024b6d88, LOCAL(num)=1xx) in new stack -- Executing [...@externaldial:2] MSet(SIP/ATA-xx-L1-024b6d88, ~~EXTEN~~=s) in new stack -- Executing [...@externaldial:3] Dial(SIP/ATA-xx-L1-024b6d88, SIP/1xxx...@link2voip-sw1,120) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx...@link2voip-sw1 -- SIP/link2voip-sw1-02477668 is making progress passing it to SIP/ATA-xx-L1-024b6d88 -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88 -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 doing dnsmgr_lookup for 'sip.ca2.link2voip.com' doing dnsmgr_lookup for 'sip.ca1.link2voip.com' == Spawn extension (ExternalDial, s, 3) exited non-zero on 'SIP/ATA-xx-L1-024b6d88' Any thoughts or ideas? If there were an error I could work on solving that, but there is none... Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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
Re: [asterisk-users] Music on Hold
The files used to be Frederic Chopin – Polonaised Op. 40-2.raw I have since replaced the raw files with the original mp3s They are now as follows: [r...@tsunami musiconhold]# ls -l . total 13320 -rw-r--r-- 1 asterisk asterisk 5415926 2009-09-16 17:18 hm1.mp3 -rw-r--r-- 1 asterisk asterisk 8217974 2009-09-16 17:18 hm2.mp3 I also have the same issue with the default files in /var/lib/asterisk/moh . On Wed, Sep 16, 2009 at 5:02 PM, Danny Nicholas da...@debsinc.com wrote: What are your actual file names (/etc/asterisk/musiconhold/Frederic Chopin – Polonaised Op. 40-2.wav?) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul *Sent:* Wednesday, September 16, 2009 4:50 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Music on Hold That was a good shot in the dark, but sadly renaming it to something simple (and removing all non ascii in the process) does not correct this. On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas da...@debsinc.com wrote: Just a “shot in the dark” but could MOH be choking on the “long file names”? (does it work on fred_chopin_pol_1)? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul *Sent:* Wednesday, September 16, 2009 4:18 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Music on Hold Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI moh show files Class: default File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1 These files were generated by SoX: Channels : 1 Sample Rate: 8000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM Endian Type: little Reverse Nibbles: no Reverse Bits : no Comment: 'Processed by SoX' This prints in the asterisk console when you attempt to put someone in hold: -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rv): == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] Goto(SIP/ATA-xx-L1-024b6d88, 1xx,1) in new stack -- Goto (phones,1xx,1) -- Executing [1xxx...@phones:1] MSet(SIP/ATA-xx-L1-024b6d88, oldcidnum=0) in new stack -- Executing [1xxx...@phones:2] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(name)=) in new stack -- Executing [1xxx...@phones:3] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(num)=xx) in new stack -- Executing [1xxx...@phones:4] Monitor(SIP/ATA-xx-L1-024b6d88, wav,/tmp/out 0 2009-09-17 03h 04m 51s CST xx,m) in new stack -- Executing [1xxx...@phones:5] Gosub(SIP/ATA-xx-L1-024b6d88, ExternalDial,s,1(1xx)) in new stack -- Executing [...@externaldial:1] MSet(SIP/ATA-xx-L1-024b6d88, LOCAL(num)=1xx) in new stack -- Executing [...@externaldial:2] MSet(SIP/ATA-xx-L1-024b6d88, ~~EXTEN~~=s) in new stack -- Executing [...@externaldial:3] Dial(SIP/ATA-xx-L1-024b6d88, SIP/1xxx...@link2voip-sw1,120) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx...@link2voip-sw1 -- SIP/link2voip-sw1-02477668 is making progress passing it to SIP/ATA-xx-L1-024b6d88 -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88 -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 doing dnsmgr_lookup for 'sip.ca2.link2voip.com' doing dnsmgr_lookup for 'sip.ca1.link2voip.com' == Spawn extension (ExternalDial, s, 3) exited non-zero on 'SIP/ATA-xx-L1-024b6d88' Any thoughts or ideas? If there were an error I could work on solving that, but there is none... Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com
Re: [asterisk-users] Music on Hold
This might be another piece of the puzzle: It would appear any application using playback functionality exits immediately. For example anything involving voicemail or playback. Phone calls work with no problem but not if asterisk must play something back. The modules are loaded however... Tsunami*CLI module show like voicemail Module Description Use Count app_voicemail.so Comedian Mail (Voicemail System) 0 I'm begining to think that the problem lies with my vendor's package. On Wed, Sep 16, 2009 at 5:30 PM, Dan Saul daniel.s...@gmail.com wrote: The files used to be Frederic Chopin – Polonaised Op. 40-2.raw I have since replaced the raw files with the original mp3s They are now as follows: [r...@tsunami musiconhold]# ls -l . total 13320 -rw-r--r-- 1 asterisk asterisk 5415926 2009-09-16 17:18 hm1.mp3 -rw-r--r-- 1 asterisk asterisk 8217974 2009-09-16 17:18 hm2.mp3 I also have the same issue with the default files in /var/lib/asterisk/moh . On Wed, Sep 16, 2009 at 5:02 PM, Danny Nicholas da...@debsinc.com wrote: What are your actual file names (/etc/asterisk/musiconhold/Frederic Chopin – Polonaised Op. 40-2.wav?) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul *Sent:* Wednesday, September 16, 2009 4:50 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Music on Hold That was a good shot in the dark, but sadly renaming it to something simple (and removing all non ascii in the process) does not correct this. On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas da...@debsinc.com wrote: Just a “shot in the dark” but could MOH be choking on the “long file names”? (does it work on fred_chopin_pol_1)? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul *Sent:* Wednesday, September 16, 2009 4:18 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Music on Hold Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI moh show files Class: default File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1 These files were generated by SoX: Channels : 1 Sample Rate: 8000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM Endian Type: little Reverse Nibbles: no Reverse Bits : no Comment: 'Processed by SoX' This prints in the asterisk console when you attempt to put someone in hold: -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rv): == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] Goto(SIP/ATA-xx-L1-024b6d88, 1xx,1) in new stack -- Goto (phones,1xx,1) -- Executing [1xxx...@phones:1] MSet(SIP/ATA-xx-L1-024b6d88, oldcidnum=0) in new stack -- Executing [1xxx...@phones:2] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(name)=) in new stack -- Executing [1xxx...@phones:3] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(num)=xx) in new stack -- Executing [1xxx...@phones:4] Monitor(SIP/ATA-xx-L1-024b6d88, wav,/tmp/out 0 2009-09-17 03h 04m 51s CST xx,m) in new stack -- Executing [1xxx...@phones:5] Gosub(SIP/ATA-xx-L1-024b6d88, ExternalDial,s,1(1xx)) in new stack -- Executing [...@externaldial:1] MSet(SIP/ATA-xx-L1-024b6d88, LOCAL(num)=1xx) in new stack -- Executing [...@externaldial:2] MSet(SIP/ATA-xx-L1-024b6d88, ~~EXTEN~~=s) in new stack -- Executing [...@externaldial:3] Dial(SIP/ATA-xx-L1-024b6d88, SIP/1xxx...@link2voip-sw1,120) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx...@link2voip-sw1 -- SIP/link2voip-sw1-02477668 is making progress passing it to SIP/ATA-xx-L1-024b6d88 -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88 -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 doing dnsmgr_lookup for 'sip.ca2.link2voip.com' doing dnsmgr_lookup for 'sip.ca1.link2voip.com' == Spawn extension (ExternalDial, s, 3) exited non-zero on 'SIP/ATA-xx-L1-024b6d88' Any thoughts or ideas? If there were an error I could work on solving that, but there is none... Thanks
Re: [asterisk-users] Music on Hold
On 17/09/09 10:40 AM, Dan Saul wrote: This might be another piece of the puzzle: It would appear any application using playback functionality exits immediately. For example anything involving voicemail or playback. Phone calls work with no problem but not if asterisk must play something back. Do you get an error in the console? What do you have in /etc/asterisk/modules.conf -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Music on hold based on user
You can use 'Set(CHANNEL(musicclass)=${MOH})' anywhere in your dialplan - so
you can set it at any stage of an inbound or outbound call (as long as it is
before the Dial/Queue command).
Eg:
[inbound]
exten = _X.,1,Set(CHANNEL(musicclass)=${MOH})
exten = _X.,n,Dial(whomever-you-want)
[outbound]
exten = _X.,1,Set(CHANNEL(musicclass)=${MOH})
exten = _X.,n,Dial(where-ever-you-want)
Then, when 'whomever-you-want' puts the call on hold - they get
'whomever-you-want's MOH.
Simples :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen
Sent: 24 July 2009 14:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Music on hold based on user
Andrew Thomas schrieb:
I do this using the setvar facility in sip.conf.
eg. setvar=MOH=music1
Then in the dialplan (extensions.conf) all you need to do is
'Set(CHANNEL(musicclass)=${MOH})'
Juan C. Crespo R. wrote:
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play
music 1, if another from Technical Support put on hold the call play
music 3, something like this
Admin - Music1
Contrallors - Music 2
Technical Support - Music 3
The way I understood the OP was that he wants different MoH classes
depending on the callee (not depending on the caller).
Philipp Kempgen
--
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] Music on hold based on user
I do this using the setvar facility in sip.conf.
eg. setvar=MOH=music1
Then in the dialplan (extensions.conf) all you need to do is
'Set(CHANNEL(musicclass)=${MOH})'
Remember, setvar in sip.conf makes that variable a global variable.
Andrew Thomas
Technical Services Manager
Juan C. Crespo R. wrote:
Hi
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play
music 1, if another from Technical Support put on hold the call play
music 3, something like this
Admin - Music1
Contrallors - Music 2
Technical Support - Music 3
Thanks
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Re: [asterisk-users] Music on hold based on user
Andrew Thomas schrieb:
I do this using the setvar facility in sip.conf.
eg. setvar=MOH=music1
Then in the dialplan (extensions.conf) all you need to do is
'Set(CHANNEL(musicclass)=${MOH})'
Juan C. Crespo R. wrote:
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play
music 1, if another from Technical Support put on hold the call play
music 3, something like this
Admin - Music1
Contrallors - Music 2
Technical Support - Music 3
The way I understood the OP was that he wants different MoH classes
depending on the callee (not depending on the caller).
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] Music on hold based on user
Here's my solution to the OP's question
- exten = s,1,answer
- exten = s,2,Set(MOH=$DB(OHCLASS/${EXTEN}))
- exten = s,3,SetMusiconhold(${MOH})
- exten = s,4,Dial(SIP/${EXTEN}),30,ikKtTm)
Just set OHCLASS exten to the value you want in advance
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: Friday, July 24, 2009 8:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Music on hold based on user
Andrew Thomas schrieb:
I do this using the setvar facility in sip.conf.
eg. setvar=MOH=music1
Then in the dialplan (extensions.conf) all you need to do is
'Set(CHANNEL(musicclass)=${MOH})'
Juan C. Crespo R. wrote:
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play
music 1, if another from Technical Support put on hold the call play
music 3, something like this
Admin - Music1
Contrallors - Music 2
Technical Support - Music 3
The way I understood the OP was that he wants different MoH classes
depending on the callee (not depending on the caller).
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] Music on hold based on user
On Thu, Jul 23, 2009 at 9:35 AM, Juan C. Crespo R.jcre...@ifxnw.com.ve wrote: Hi Guys I wonder if its possible to set a different MoH based on groups, I mean if one of the Admin group put on hold the call play music 1, if another from Technical Support put on hold the call play music 3, something like this Admin - Music1 Contrallors - Music 2 Technical Support - Music 3 Seems like a perfect use of SetMusicOnHold.. http://www.voip-info.org/wiki/view/Asterisk+cmd+SetMusicOnHold -jonathan ___ -- 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] Music on hold based on user
Juan C. Crespo R. schrieb:
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play music
1, if another from Technical Support put on hold the call play music 3,
something like this
Admin - Music1
Contrallors - Music 2
Technical Support - Music 3
Some dialplan logic around Set(CHANNEL(musicclass)=...) should do
the trick I guess.
Maybe the easiest way (in Asterisk 1.6) would be to add
setvar=musicclass=admin / setvar=musicclass=support / ...
to your SIP peers and then do something like
_X. = {
Set(CHANNEL(musicclass)=${SIPPEER(${EXTEN},chanvar[musicclass])})
Dial(SIP/${EXTEN});
}
in your dialplan (untested).
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] Music on hold based on user
Depending on how your dialplan is set you can use the SetMusicOnHold application after creating classes in your musiconhold.conf http://www.asteriskguru.com/tutorials/setmusiconhold.html Ish Juan C. Crespo R. wrote: Hi Guys I wonder if its possible to set a different MoH based on groups, I mean if one of the Admin group put on hold the call play music 1, if another from Technical Support put on hold the call play music 3, something like this Admin - Music1 Contrallors - Music 2 Technical Support - Music 3 Thanks -- ___ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- 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] Music on Hold
Julien Claassen wrote: Hello! I've configured Music on Hold in asterisk, the only, most certainly, stupid problem I have is, which DTMFs to send to activate and deactivate it. If I use the cli, I can establish a call with originate. With the misdn send digit command I can send a number of digits to the other party. But what are the combinations to put the other one on hold? Or do I have to use a completely different mechanism? Any help here is appreciated. A pointer to the right part of the documentation is completely sufficient. Warm regards Julien Putting a person on hold using DTMF is part of the feature code mechanism. You configure it in features.conf. ___ -- 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] Music on Hold
Thanks Brent! I'll have a look there in features.conf. Warm regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] music on hold file formats
On Tue, Jun 23, 2009 at 5:40 PM, Ron nha...@gmail.com wrote: I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. That could be for any number of reasons, including a overly lossy mp3 to begin with. the mp3 moh is very clear when i use g711, only on g729 i'm having the issue, could that be an issue of lossy MP3 still? Ron, Keep in mind that G.729a is a low bit-rate CODEC optimized for speech compression. It does not do well with music, or even DTMF tones. G.711, on the other hand, has greater dynamic range, so music sounds better. It would be nice if Asterisk had an option like CCM that forces MoH to use G.711, while the voice calls still use G.729. Sincerely, Trevor Hammonds ___ -- 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] music on hold file formats
On Tue, 23 Jun 2009, Ron wrote: Hi, what software do i need to convert an mp3 to a g729 format? I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. so what i'd like to do is when they upload an MP3 i will make a copy on g729 format, so that asterisk can choose which file to play depending on what codec is being used by the user. Have you tried the convert command inside Asterisk? (1.4+) 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] music on hold file formats
On Tue, Jun 23, 2009 at 3:31 AM, Ronnha...@gmail.com wrote: Hi, what software do i need to convert an mp3 to a g729 format? I'm not aware of a package to do it in one step. Sox can work with a large number of formats, but mp3 isn't one of them. I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. That could be for any number of reasons, including a overly lossy mp3 to begin with. so what i'd like to do is when they upload an MP3 i will make a copy on g729 format, so that asterisk can choose which file to play depending on what codec is being used by the user. Why did you choose mp3 in the first place? How about turn this around on the customer and have them upload a g729 file? ___ -- 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] music on hold file formats
David Backeberg wrote: I'm not aware of a package to do it in one step. Sox can work with a large number of formats, but mp3 isn't one of them. Yes, sox can work with MP3 files, but not G.729. Not all distributions include MP3 support in their sox builds due to patent licensing concerns, though. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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] music on hold file formats
David Backeberg wrote: On Tue, Jun 23, 2009 at 3:31 AM, Ronnha...@gmail.com wrote: Hi, what software do i need to convert an mp3 to a g729 format? I'm not aware of a package to do it in one step. Sox can work with a large number of formats, but mp3 isn't one of them. hi sir, i'm ok if there's two step, may i know what softwares are used for those steps? preferably conversion is via CLI I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. That could be for any number of reasons, including a overly lossy mp3 to begin with. the mp3 moh is very clear when i use g711, only on g729 i'm having the issue, could that be an issue of lossy MP3 still? so what i'd like to do is when they upload an MP3 i will make a copy on g729 format, so that asterisk can choose which file to play depending on what codec is being used by the user. Why did you choose mp3 in the first place? How about turn this around on the customer and have them upload a g729 file? my users are not technically knowledgeable on different file formats and mp3's and wav are mostly what the users know. when they upload an MP3 i would like to convert it and save it on different formats (e.g. g729 and g711). thanks regards ron w.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] music on hold file formats
http://www.voip-info.org/wiki/view/sox On Tue, Jun 23, 2009 at 8:40 PM, Ronnha...@gmail.com wrote: David Backeberg wrote: On Tue, Jun 23, 2009 at 3:31 AM, Ronnha...@gmail.com wrote: Hi, what software do i need to convert an mp3 to a g729 format? I'm not aware of a package to do it in one step. Sox can work with a large number of formats, but mp3 isn't one of them. hi sir, i'm ok if there's two step, may i know what softwares are used for those steps? preferably conversion is via CLI I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. That could be for any number of reasons, including a overly lossy mp3 to begin with. the mp3 moh is very clear when i use g711, only on g729 i'm having the issue, could that be an issue of lossy MP3 still? so what i'd like to do is when they upload an MP3 i will make a copy on g729 format, so that asterisk can choose which file to play depending on what codec is being used by the user. Why did you choose mp3 in the first place? How about turn this around on the customer and have them upload a g729 file? my users are not technically knowledgeable on different file formats and mp3's and wav are mostly what the users know. when they upload an MP3 i would like to convert it and save it on different formats (e.g. g729 and g711). thanks regards ron w.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] music on hold file formats
On Wed, 24 Jun 2009, Ron wrote: David Backeberg wrote: On Tue, Jun 23, 2009 at 3:31 AM, Ronnha...@gmail.com wrote: what software do i need to convert an mp3 to a g729 format? I'm not aware of a package to do it in one step. Sox can work with a large number of formats, but mp3 isn't one of them. Sox can if it is compiled in. It isn't in CentOS. hi sir, i'm ok if there's two step, may i know what softwares are used for those steps? preferably conversion is via CLI I use: ) mplayer (if available) or file to determine if the file is an MP3. ) mpg123 to convert from MP3 to WAV (if needed). ) normalize so everything sounds about the same loudness. ) sox to make sure the channel count, sample size, signedness and sampling rate are correct. I'm only interested in WAV (-c 1 -s -r 8000 -w), SLN (-c 1 -r 8000 -s -t raw -w), and ULAW (-c 1 -r 8000 -t ul). I wrap them all up in a script named transcode.sh to make it easy for me and my users to convert any old bunch of files with a single command. Sox can't do g729, but Asterisk can using the CLI command file convert file_in.format file_out.format so you could extend the concept. my users are not technically knowledgeable on different file formats and mp3's and wav are mostly what the users know. when they upload an MP3 i would like to convert it and save it on different formats (e.g. g729 and g711). thanks If you get a choice, take the WAV and convert it to whatever formats you need -- skip MP3 altogether. Thanks in advance, Steve Edwards sedwa...@sedwards.com 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] music on hold using mms
I did have a lot of problems with mpg123 some time ago but it seems to behave pretty well at this point. I suppose you could use something other than mpg123 as long as it was capable of processing the stream. Rilawich Ango wrote: Thanks. But I heard that mpg123 uses much resources (CPU memory) of each connection. Is it true? How about using madplay? On 4/28/09, M Hulber asterisk-ad...@hulber.com wrote: Didn't do mms but have implemented using Shoutcast. I have instructions at the link below: http://mark.hulber.com/voip/configuration/shoutcast-musiconhold-in-asterisk-16/ Rilawich Ango wrote: Hi, I follow the web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf - mohstream.sh , to configure music on hold to play using mms but failed. Anyone can play using mms? ango ___ -- 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 -- MARK. Hulber Technologies asterisk-ad...@hulber.com Read my blog : http://mark.hulber.com Follow @hulber on Twitter: http://twitter.com/hulber ___ -- 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 -- MARK. Hulber Technologies asterisk-ad...@hulber.com Read my blog : http://mark.hulber.com Follow @hulber on Twitter: http://twitter.com/hulber ___ -- 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] music on hold using mms
Didn't do mms but have implemented using Shoutcast. I have instructions at the link below: http://mark.hulber.com/voip/configuration/shoutcast-musiconhold-in-asterisk-16/ Rilawich Ango wrote: Hi, I follow the web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf - mohstream.sh , to configure music on hold to play using mms but failed. Anyone can play using mms? ango ___ -- 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 -- MARK. Hulber Technologies asterisk-ad...@hulber.com Read my blog : http://mark.hulber.com Follow @hulber on Twitter: http://twitter.com/hulber ___ -- 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] music on hold using mms
Thanks. But I heard that mpg123 uses much resources (CPU memory) of each connection. Is it true? How about using madplay? On 4/28/09, M Hulber asterisk-ad...@hulber.com wrote: Didn't do mms but have implemented using Shoutcast. I have instructions at the link below: http://mark.hulber.com/voip/configuration/shoutcast-musiconhold-in-asterisk-16/ Rilawich Ango wrote: Hi, I follow the web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf - mohstream.sh , to configure music on hold to play using mms but failed. Anyone can play using mms? ango ___ -- 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 -- MARK. Hulber Technologies asterisk-ad...@hulber.com Read my blog : http://mark.hulber.com Follow @hulber on Twitter: http://twitter.com/hulber ___ -- 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] music-on-hold kicks in and disconnects/interrupt the call
Joseph wrote: I'm using Asterisk 1.4.22.1 When I'm on active call it happens many times the call gets interrupted by music-on-hold without my pressing any button. MOH just kicks in and int erupt the call and I have no way of getting the call back. Did anybody experienced anything like this? No - do you have any dialplan code or cli output to show for this excitement? PaulH ___ -- 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] Music on Hold doesn't play back for external callers
Do you have zaptel timing working ? (or dahdi) ? If no timing is available then if there's no incoming audio frames coming from the external SIP channel then no outgoing audio will be produced (even if you have MOH application working) The trigger to fire the outgoing audio frame comes from the incoming audio frame. So check if the Answer + Echo application work. Also if you have VAD which means your SIP device doesn't send audio frames when there's silence detected = you'd also have the same problem = not see audio going to your phone. Martin On Fri, Mar 20, 2009 at 6:07 PM, Greg Hinson greg_hin...@msn.com wrote: Hey all; I am experiencing an issue with music on Hold. I am running asterisk version 1.4.22, and have a default script set up in two places for MoH playback. For internal devices to my network that are SIP peering with asterisk, they simply dial 123 and hear the MoH music immediately. I'm using the default setup, where it just plays the wave files in the /var/lib/asterisk/moh directory. I also have in my General settings in sip.conf to use the default musicclass. *On-net context script for MoH in extensions.conf:* exten = 123,1,Answer() exten = 123,2,MusicOnHold(default) *Calls from the PSTN are matching this Moh setup in extensions.conf:* exten = _1720XXX,1,Answer() exten = _1720XXX,2,MusicOnHold(default) *Output captured when internal devices attempt to dial number associated with MoH:(MoH is heard)* -- Executing [...@home:1] Answer(SIP/10.1.1.254-b77089b0, ) in new stack -- Executing [...@home:2] MusicOnHold(SIP/10.1.1.254-b77089b0, default) in new stack -- Started music on hold, class 'default', on SIP/10.1.1.254-b77089b0 *Output captured when external callers dial into asterisk from the PSTN, they get dead air instead of MoH:* -- Executing [1720...@home:1] Answer(SIP/1720XXX-b7704a38, ) in new stack -- Executing [1720...@home:2] MusicOnHold(SIP/1720XXX-b7704a38, default) in new stack -- Started music on hold, class 'default', on SIP/1720XXX-b7704a38 You see rule 1(answer) and rule 2(moh) being hit on both call attempts, but the external caller continues to get dead air. has anyone seen this one before? The internal devices requesting MoH are using the SCCP protocol load image(registered elsewhere) and then they are SIP trunked to Asterisk. Asterisk uses a second separate SIP trunk to peer with the PSTN. -Greg -- Windows Live™ SkyDrive: Get 25 GB of free online storage. Check it out.http://windowslive.com/online/skydrive?ocid=TXT_TAGLM_WL_skydrive_032009 ___ -- 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] Music On Hold
In my situation AMI is not an option. When somebdy puts a call on hold, on asterisk console I can see messages like Started music on hold, class 'default', on SIP/ and Started music on hold, class 'default', on SIP/. I guess the only way in my scenerio is to modify res_musiconhold.so. Thanks for your help. Idris -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ex Vito Sent: Monday, February 02, 2009 5:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music On Hold On Fri, Jan 30, 2009 at 3:23 PM, Danny Nicholas da...@debsinc.com wrote: The dialplan AFAIK doesn't cover HOLD handling. If you can spare the overhead, you can make a daemon to watch hints and run a script whenever the hint for a line goes to hold and changes from hold to inuse. Just run asterisk –rx core show hints and asterisk –rx core show channels and integrate the 2 outputs. For your purpose, you can probably just use the first command. You should instaed use the AMI and create an event based solution instead of relying on polling via asterisk -rx !... Check out: http://www.voip-info.org/wiki-Asterisk+manager+API http://www.voip-info.org/wiki/view/asterisk+manager+events Cheers, -- 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 ___ -- 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] Music On Hold
On Mon, Feb 2, 2009 at 8:39 AM, Idris AVCI idris.a...@vodatech.com.tr wrote: In my situation AMI is not an option. When somebdy puts a call on hold, on asterisk console I can see messages like Started music on hold, class 'default', on SIP/ and Started music on hold, class 'default', on SIP/. I guess the only way in my scenerio is to modify res_musiconhold.so. ...and for each of those console messages an AMI event is fired; it should be relatively simple to attach script execution to those events. Also, I confess that I'm curious as to what environment you're running where using AMI is not an option but hacking res_musiconhold.so is... Good luck, anyway. -- 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] Music On Hold
On Fri, Jan 30, 2009 at 3:23 PM, Danny Nicholas da...@debsinc.com wrote: The dialplan AFAIK doesn't cover HOLD handling. If you can spare the overhead, you can make a daemon to watch hints and run a script whenever the hint for a line goes to hold and changes from hold to inuse. Just run asterisk –rx core show hints and asterisk –rx core show channels and integrate the 2 outputs. For your purpose, you can probably just use the first command. You should instaed use the AMI and create an event based solution instead of relying on polling via asterisk -rx !... Check out: http://www.voip-info.org/wiki-Asterisk+manager+API http://www.voip-info.org/wiki/view/asterisk+manager+events Cheers, -- 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] Music On Hold
The dialplan AFAIK doesn't cover HOLD handling. If you can spare the overhead, you can make a daemon to watch hints and run a script whenever the hint for a line goes to hold and changes from hold to inuse. Just run asterisk -rx core show hints and asterisk -rx core show channels and integrate the 2 outputs. For your purpose, you can probably just use the first command. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Idris AVCI Sent: Friday, January 30, 2009 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Music On Hold Hi, I am using asterisk version 1.4.22.1 on a centos 5.2 machine. Is there any way to run a script somebody puts the call on/off hold ? The script must be run both on hold and off hold. Best ragards. Idris ___ -- 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] music on hold
See: http://astrecipes.net/index.php?q=AstRecipes/Music-on-hold%20without%20MPG123 ___ -- 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] music on hold
On Tue, Nov 11, 2008 at 05:53:39PM +1100, Lee, John (Sydney) wrote: The reason is your audio file is too high quality. Asterisk can only play back audio file of 4000Hz. 8000Hz (and also: mono, 16 bit sample rate). What is the output of: file path/to/sound.wav -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] music on hold
^_^ asterisk should Encoding voice in 8KHZ ,16k bits,mono8 ,i have formate the .wav files. thank you for your advice,best regard. 2008-11-11 邱磊 发件人： Peter Evans 发送时间： 2008-11-11 14:57:36 收件人： Asterisk Users Mailing List - Non-Commercial Discussion 抄送： 主题： Re: [asterisk-users] music on hold On Tue, Nov 11, 2008 at 02:35:19PM +0800, 邱磊 wrote: hii guys: i get the message from the asterisk: Started music on hold, class 'default', on Local/[EMAIL PROTECTED],1 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav [2008-11-11 14:32:41] WARNING[1781]: res_musiconhold.c:259 ast_moh_files_next: Unable to open file '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory -- Stopped music on hold on Local/[EMAIL PROTECTED],1 TOM SKYPE + 163.com = I guess your first language is mandarin. You know, life would be so much easier if people would read error messages and think for a moment. Undoubtably, there are times when the error message is cryptic beyond mortal understanding and you need to summon an elder thing to help you read it, but this one is not of that ilk. '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory This could be a tough one. If you can't solve this without help, should you really be playing with ancient scrolls of wisdom in the first place? P ps: 没有??的文件或目? but that probably doesnt work because I am using a Japanese font ^^; ___ -- 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] music on hold
thank you，Ihave get the solution。 asterisk should Encoding voice in 8KHZ ,16k bits,mono8 ^^ 2008-11-11 邱磊 发件人： Lee, John (Sydney) 发送时间： 2008-11-11 14:55:00 收件人： Asterisk Users Mailing List - Non-Commercial Discussion 抄送： 主题： Re: [asterisk-users] music on hold The reason is your audio file is too high quality. Asterisk can only play back audio file of 4000Hz. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ?? Sent: Tuesday, 11 November 2008 5:35 PM To: asterisk-users Subject: [asterisk-users] music on hold hii guys: i get the message from the asterisk: Started music on hold, class 'default', on Local/[EMAIL PROTECTED],1 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav [2008-11-11 14:32:41] WARNING[1781]: res_musiconhold.c:259 ast_moh_files_next: Unable to open file '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory -- Stopped music on hold on Local/[EMAIL PROTECTED],1 how can i solve the issue? thanks 2008-11-11 $Bn9b}(J ___ -- 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] music on hold
On Tue, 11 Nov 2008, Peter Evans wrote: On Tue, Nov 11, 2008 at 02:35:19PM +0800, 邱磊 wrote: hii guys: i get the message from the asterisk: Started music on hold, class 'default', on Local/[EMAIL PROTECTED],1 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav [2008-11-11 14:32:41] WARNING[1781]: res_musiconhold.c:259 ast_moh_files_next: Unable to open file '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory -- Stopped music on hold on Local/[EMAIL PROTECTED],1 TOM SKYPE + 163.com = I guess your first language is mandarin. You know, life would be so much easier if people would read error messages and think for a moment. Undoubtably, there are times when the error message is cryptic beyond mortal understanding and you need to summon an elder thing to help you read it, but this one is not of that ilk. '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory This could be a tough one. If you can't solve this without help, should you really be playing with ancient scrolls of wisdom in the first place? Isn't it horribly embarrasing when you publicly trash someone and are WRONG? Why do people feel the need to be so cruel in the first place? j ___ -- 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] music on hold
The reason is your audio file is too high quality. Asterisk can only play back audio file of 4000Hz. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ?? Sent: Tuesday, 11 November 2008 5:35 PM To: asterisk-users Subject: [asterisk-users] music on hold hii guys: i get the message from the asterisk: Started music on hold, class 'default', on Local/[EMAIL PROTECTED],1 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav [2008-11-11 14:32:41] WARNING[1781]: res_musiconhold.c:259 ast_moh_files_next: Unable to open file '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory -- Stopped music on hold on Local/[EMAIL PROTECTED],1 how can i solve the issue? thanks 2008-11-11 邱磊 ___ -- 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] Music on hold for sub tenants
Yes, you can set moh in sip.conf or zapata.conf. The options are mohinterpret= mohsuggest=. I think last time I used them (1.2.x) they were just moh= but it seems mohsuggest=class will do what you want it to. On Sat, Oct 4, 2008 at 2:57 PM, carl Lougher [EMAIL PROTECTED] wrote: This seems to be related to inbound calls. So would this work for music on transfers within that context as well as hitting the hold key on calls? ___ -- 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
Re: [asterisk-users] Music on hold for sub tenants
This seems to be related to inbound calls. So would this work for music on transfers within that context as well as hitting the hold key on calls? --- On Fri, 26/9/08, Darrick Hartman [EMAIL PROTECTED] wrote: From: Darrick Hartman [EMAIL PROTECTED] Subject: Re: [asterisk-users] Music on hold for sub tenants To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, 26 September, 2008, 4:52 AM ...since everyone else top posted. Take a look at the application setmusiconhold. CLI core show application SetMusicOnHold You can use this in a dialplan as follows: [tenant1incoming] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(tenant1sounds/welcome) exten = s,n,SetMusicOnHold(tenant1) [tenant2incoming] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(tentant2sounds/welcome) exten = s,n,SetMusicOnHold(tenant2) Use that with the previously supplied info. Darrick carl Lougher wrote: Hi, I tried this but it still uses the default moh. Is there some way to define it based on a context in the sip.conf or extensions.conf??? Taff... --- On Fri, 26/9/08, Nhadie [EMAIL PROTECTED] wrote: From: Nhadie [EMAIL PROTECTED] Subject: Re: [asterisk-users] Music on hold for sub tenants To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, 26 September, 2008, 4:10 AM Hi, i think you can define it like this: [moh-company-a] mode=files directory=/var/lib/asterisk/moh/companya [moh-company-b] mode=files directory=/var/lib/asterisk/moh/companyb regards, nhadie carl Lougher wrote: Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B ___ -- 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
Re: [asterisk-users] Music on hold for sub tenants
Hi, i think you can define it like this: [moh-company-a] mode=files directory=/var/lib/asterisk/moh/companya [moh-company-b] mode=files directory=/var/lib/asterisk/moh/companyb regards, nhadie carl Lougher wrote: Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B Any help much appreciated.. Thanks, Taff... ___ -- 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
Re: [asterisk-users] Music on hold for sub tenants
Hi, I tried this but it still uses the default moh. Is there some way to define it based on a context in the sip.conf or extensions.conf??? Taff... --- On Fri, 26/9/08, Nhadie [EMAIL PROTECTED] wrote: From: Nhadie [EMAIL PROTECTED] Subject: Re: [asterisk-users] Music on hold for sub tenants To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, 26 September, 2008, 4:10 AM Hi, i think you can define it like this: [moh-company-a] mode=files directory=/var/lib/asterisk/moh/companya [moh-company-b] mode=files directory=/var/lib/asterisk/moh/companyb regards, nhadie carl Lougher wrote: Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B Any help much appreciated.. Thanks, Taff... ___ -- 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 ___ -- 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
Re: [asterisk-users] Music on hold for sub tenants
...since everyone else top posted. Take a look at the application setmusiconhold. CLI core show application SetMusicOnHold You can use this in a dialplan as follows: [tenant1incoming] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(tenant1sounds/welcome) exten = s,n,SetMusicOnHold(tenant1) [tenant2incoming] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(tentant2sounds/welcome) exten = s,n,SetMusicOnHold(tenant2) Use that with the previously supplied info. Darrick carl Lougher wrote: Hi, I tried this but it still uses the default moh. Is there some way to define it based on a context in the sip.conf or extensions.conf??? Taff... --- On Fri, 26/9/08, Nhadie [EMAIL PROTECTED] wrote: From: Nhadie [EMAIL PROTECTED] Subject: Re: [asterisk-users] Music on hold for sub tenants To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, 26 September, 2008, 4:10 AM Hi, i think you can define it like this: [moh-company-a] mode=files directory=/var/lib/asterisk/moh/companya [moh-company-b] mode=files directory=/var/lib/asterisk/moh/companyb regards, nhadie carl Lougher wrote: Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B ___ -- 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
Re: [asterisk-users] music on hold is not working
I have made class for MOH uploaded a mp3 file to the folder. Now I am using this class for music on hold during dialing. Now when call has been established, I put the other end on hold. So from that end I should listen uploaded file. But I am not getting audio. From memory, you need to install asterisk-addons in order to play mp3 file. The default audio file is .gsm ___ -- 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
Re: [asterisk-users] Music on hold
Vazquez David wrote:
Hi,
I'm getting this bizarre problem. Whenever I dial (through misdn) and
try to listen to my music on hold, I get this:
-- Started music on hold, class 'default', on channel 'mISDN/3-u72'
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 6644 requested bytes on mISDN/3-u72
-- Stopped music on hold on mISDN/3-u72
Any idea???
Thanks :D
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Solved :
I didn't have an answer statement in my extensions.conf
The working context:
exten = 03,1,Answer()
exten = 03,2,Queue(${EXTEN})
exten = 03,3,Hangup()
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