[Asterisk-Users] Music on Hold

[Ernest W. Lessenger]

Does anybody have a good source for hold music? I can see a number of 
companies on the web that sell royalty-free MOH, but they don't all provide 
samples. The customer service desk has requested "calming, not sleeping, 
but calming" and "this is a high-tech company, so make it 'techie' [sic]".

Thanks,
--Ernest
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[Asterisk-Users] Music on hold...

[Chris Hariga]

Hi,

I need a sound card and mpg123 for music on hold??? When I call Digium
the guys toll me "is not necessary to have a sound card". My music on
hold doesn't work :((

Best regards,

Chris HARIGA
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Thursday, October 16, 2003 8:24 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SER vs STUND with Asterisk..

Olle E. Johansson wrote:

> WipeOut wrote:
>
>> Olle E. Johansson wrote:
>>
>>> WipeOut wrote:
>>>
 Anyway, I decided to go and have a quick read through the SER docs 
 and in the section about NAT they say that the best way to address 
 NAT is to use STUN or uPNP..
>>>
>>>
>>> STUN is helpful, but as I understand it analyzes the situation and 
>>> reports
>>> the configuration of a NAT. It doesn't help you keeping the NAT 
>>> session open,
>>> as SER module nathelper or the FWD/Jasomi solution.
>>> Check here http://www.voip-info.org/wiki-SER+module+nathelper
>>> It's ugly, but what it does is sending UDP packets from the outside 
>>> to the
>>> NAT to keep the ports open for incoming calls. NAT is an ugly thing,
>>> so it propably needs ugly solutions... ;-) 
>>
>>
>> Looking at that page you mentioned it still seems to me that the 
>> "nathelper" module for SER and adding nat=yes to the sip.conf 
>> essentially do the same thing apart from the "NAT pings" you 
>> mentioned below..
>
> Right. There's also more commands so that you can tweak SER into doing
> different kinds of SIP message mangling than the - still rather 
> undocumented -
> nat=yes. My guess is that nat=yes changes the Contact to the actual IP

> used
> to contact Asterisk, not the IP given in the SIP headers. Right? 

Not sure about the intimate details of what nat=yes does exactly but it 
defiantely works, also have just found out (thanks to John Todd) the if 
you add "qualify=500" to your UA configuration in the sip.conf then it 
essentially uses keep alives in the form of a OPTIONS request every 60 
seconds.. So by having nat= and qualify= removes the need to have SER 
and the nathelper module.. (No doubt there is more that SER can do and 
if you really need those features then go for it..)

>
>>> As I understand it, it works like this:
>>> * Client on the inside of a NAT registers to an outside SIP Proxy
>>> * THe outside SIP Proxy keeps sending UDP packets ("NAT PINGS") to
the
>>>   client to keep the UDP session open in the NAT
>>> * When someone calls, the session is open and the client (UAC/S) may
>>>   answer...
>>> * In addition to the solution for handling SIP this way, there's a
>>>   need for an RTP media server to handle the RTP stream.
>>
>> I guess that if you use SER or STUN and Asterisk the RTP is still 
>> going to be an issue if the call is needing to go between two SIP 
>> UA's that are both behind NAT (UA---NAT--Internet--NAT--UA) so the 
>> RTP streams are going to have to go via the central server (aka 
>> canreinvite=no in Asterisk).. So if NAT is in the picture you have no

>> choice but to load the server with all the traffic..
>
> Right. That's where the PortaOne RTP proxy - or Asterisk - come in.
> The RTP proxy in combination with SERs nathelper changes the SDP to
> point to the RTP proxy in this case and informs the RTP proxy of the
> session through a Unix pipe.

Personally I think I would stick with Asterisk to handle all the RTP 
traffic, just by adding canreinvite=no to the sip.conf will cause all 
traffice between the endpoints to go via Asterisk.. The fewer systems 
that need to be tied together the better IMO.. If it can all be done 
with one then there is less to go wrong.. :)

>
 So my question is would it not be better to couple STUND 
 (Vovida.org) with Asterisk and then use nat=yes in the sip.conf for

 UA's that do not support STUN, instead of using SER which would be 
 like learning Asterisk all over again and would require you to 
 learn how to use the SER config language to manage your NAT 
 transtaltions..
>>>
>>> Integrating a STUN server into ASterisk... I don't see the point. 
>>> But if
>>> you're talking about asterisk as a SIP client (registrering to other

>>> SIP
>>> servers) supporting STUN to find out if it's behind a NAT and how
the
>>> NAT works, yes, that's a good idea.
>>
>> I wasn't talking about intergrating STUN into asterisk, I was 
>> thinking more along the lines of using STUND in conjunction with 
>> Asterisk instead of SER and Asterisk.. :)
>
> Sorry, my misunderstanding. Are you thinking the way I did, with
Asterisk
> as a SIP client or are you thinking of supporting Asterisk's SIP
clients,
> the phones, with a STUND? 

I was thinking of the supporting the SIP clients (phones).. I think that

it is the resposibility of the server to handle as much complexity as 
possible making it easier for the UA's to be configured.. So if you are 
trying to connect Asterisk(as a client) to a third party to route your 
calls I would

[Asterisk-Users] Music on hold...

[CW_ASN]

No, you don't need a sound card.
Do you have ztdummy loaded or zaptel device in your system?

Regards,

Gus

- Original Message - 
From: "Chris Hariga" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, October 19, 2003 8:19 PM
Subject: [Asterisk-Users] Music on hold...


> Hi,
> 
> I need a sound card and mpg123 for music on hold??? When I call Digium
> the guys toll me "is not necessary to have a sound card". My music on
> hold doesn't work :((
> 
> Best regards,
> 
> Chris HARIGA
>  
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
> Sent: Thursday, October 16, 2003 8:24 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] SER vs STUND with Asterisk..
> 
> Olle E. Johansson wrote:
> 
> > WipeOut wrote:
> >
> >> Olle E. Johansson wrote:
> >>
> >>> WipeOut wrote:
> >>>
> >>>> Anyway, I decided to go and have a quick read through the SER docs 
> >>>> and in the section about NAT they say that the best way to address 
> >>>> NAT is to use STUN or uPNP..
> >>>
> >>>
> >>> STUN is helpful, but as I understand it analyzes the situation and 
> >>> reports
> >>> the configuration of a NAT. It doesn't help you keeping the NAT 
> >>> session open,
> >>> as SER module nathelper or the FWD/Jasomi solution.
> >>> Check here http://www.voip-info.org/wiki-SER+module+nathelper
> >>> It's ugly, but what it does is sending UDP packets from the outside 
> >>> to the
> >>> NAT to keep the ports open for incoming calls. NAT is an ugly thing,
> >>> so it propably needs ugly solutions... ;-) 
> >>
> >>
> >> Looking at that page you mentioned it still seems to me that the 
> >> "nathelper" module for SER and adding nat=yes to the sip.conf 
> >> essentially do the same thing apart from the "NAT pings" you 
> >> mentioned below..
> >
> > Right. There's also more commands so that you can tweak SER into doing
> > different kinds of SIP message mangling than the - still rather 
> > undocumented -
> > nat=yes. My guess is that nat=yes changes the Contact to the actual IP
> 
> > used
> > to contact Asterisk, not the IP given in the SIP headers. Right? 
> 
> Not sure about the intimate details of what nat=yes does exactly but it 
> defiantely works, also have just found out (thanks to John Todd) the if 
> you add "qualify=500" to your UA configuration in the sip.conf then it 
> essentially uses keep alives in the form of a OPTIONS request every 60 
> seconds.. So by having nat= and qualify= removes the need to have SER 
> and the nathelper module.. (No doubt there is more that SER can do and 
> if you really need those features then go for it..)
> 
> >
> >>> As I understand it, it works like this:
> >>> * Client on the inside of a NAT registers to an outside SIP Proxy
> >>> * THe outside SIP Proxy keeps sending UDP packets ("NAT PINGS") to
> the
> >>>   client to keep the UDP session open in the NAT
> >>> * When someone calls, the session is open and the client (UAC/S) may
> >>>   answer...
> >>> * In addition to the solution for handling SIP this way, there's a
> >>>   need for an RTP media server to handle the RTP stream.
> >>
> >> I guess that if you use SER or STUN and Asterisk the RTP is still 
> >> going to be an issue if the call is needing to go between two SIP 
> >> UA's that are both behind NAT (UA---NAT--Internet--NAT--UA) so the 
> >> RTP streams are going to have to go via the central server (aka 
> >> canreinvite=no in Asterisk).. So if NAT is in the picture you have no
> 
> >> choice but to load the server with all the traffic..
> >
> > Right. That's where the PortaOne RTP proxy - or Asterisk - come in.
> > The RTP proxy in combination with SERs nathelper changes the SDP to
> > point to the RTP proxy in this case and informs the RTP proxy of the
> > session through a Unix pipe.
> 
> Personally I think I would stick with Asterisk to handle all the RTP 
> traffic, just by adding canreinvite=no to the sip.conf will cause all 
> traffice between the endpoints to go via Asterisk.. The fewer systems 
> that need to be tied together the better IMO.. If it can all be done 
> with one then there is less to go wrong.. :)
> 
> >
> >>>> So my question is would it not be better to

[Asterisk-Users] music on hold`

[mick]

when I put a station on hold I receive this message


res_musiconhold.c, Line 280 (monmp3thread): Read 372 bytes of audio
while expecting 1600




Regards Mick

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[Asterisk-Users] Music on Hold

[Phillip Jackson, Director of IT]

Having a weird issue with on hold music ... I do have mpg123 installed.

When requesting extension  for testing, which is setup as:

exten => ,1,Answer  ; Answer the line
exten => ,2,DigitTimeout,5  ; Set Digit Timeout to 5 seconds
exten => ,3,MP3Player(${MP3ROOT}/sample-hold.mp3)

I recieve this err:

-- Executing MP3Player("SIP/100-26af", "/sample-hold.mp3") in new stack
WARNING[1217602880]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error:
Resource temporarily unavailable
NOTICE[1217602880]: File app_mp3.c, Line 80 (timed_read): Selected timed
out/errored out with 0

Not sure what's up...

Phillip

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[Asterisk-Users] music on hold

[zoa]

Sometimes when people hang up, or the call gets interrupted for some 
reason, music on hold starts playing. (i use app_dial without extra 
parameters and no moh is set in extensions.conf)

Any suggestions on how i could get rid of this 'feature' ?

greetz,

zoa.

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[Asterisk-Users] Music on hold

[Jeremy Bogan]

Hi,

I can't seem to get any of my own MP3's to work with Asterisk as music 
on hold. The default sample one works fine, but if I place another one 
in there, Asterisk fails to start. I have removed all ID3 tag 
information and also made it 128kbps mono. Is there something i'm 
missing?

Thanks!

--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
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[asterisk-users] Music on hold

[Fons van der Beek]

Because i want a ringing signal while people are in a waiting queue i've 
created a wav file containing our local ringing indication
If I make an inside call to the queue, the correct sound is played, but 
when i make an external call, no signal is heard.
everything else looks ok, and all other functions are ok

Can somebody point me out what i could have done wrong?

my musiconhold.conf
---
[ringing]
mode=files
directory=/var/lib/asterisk/ringing

i've created the wav file using
sox ringing.wav -r 8000 -c 1 -s -w ringingmono.wav resample -ql

This file is located at /var/lib/asterisk/ringing

my queues.conf

[receptie]
musicclass = ringing
strategy = ringall
timeout = 300
retry = 5
member => SIP/202
member => SIP/227

the CLI shows..
---

-- Executing [EMAIL PROTECTED]:1] Answer("SIP/04757690XX-08ef8ee8", "") in 
new stack
-- Executing [EMAIL PROTECTED]:2] LookupCIDName("SIP/0475769002-08ef8ee8", 
"") in new stack
[Feb 17 13:18:10] WARNING[28267]: app_lookupcidname.c:72 
lookupcidname_exec: LookupCIDName is deprecated.  Please use 
${DB(cidname/${CALLERID(num)})} instead.
-- Executing [EMAIL PROTECTED]:3] Queue("SIP/04757690XX-08ef8ee8", 
"receptie") in new stack
-- Started music on hold, class 'ringing', on SIP/04757690XX-08ef8ee8
-- SIP/227-08efce58 is ringing
-- SIP/201-08f06f68 is ringing
-- SIP/414-08f04930 is ringing
-- SIP/201-08f06f68 is ringing
-- SIP/201-08f06f68 is ringing
-- SIP/201-08f06f68 is ringing
-- SIP/201-08f06f68 answered SIP/04757690XX-08ef8ee8
-- Stopped music on hold on SIP/0475769002-08ef8ee8





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[asterisk-users] Music On Hold

[Joel Hill]

Hi All,

I need to have the same file played from MoH every time someone gets to
MoH from a Dial. I want to play marketing messages from it and I want it
to start from file 1 every time.

Anyone know if/how this can be done?

Cheers,

Joel.


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[asterisk-users] Music On Hold

[itgasterisk]

Hello everyone,

I am having a bit of problem getting MusicOnhold to play.

I am running Asterisk 1.4 with MPG123 0.59 installed.

And here's what i see in the debugging window of asterisk:

-- Started music on hold, class 'default', on channel 
'SIP/x123-082043d0'
-- Stopped music on hold on SIP/x123-082043d0

Any idea why it is not playing the file at all?

thanks

Eric


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[asterisk-users] music on hold

[邱磊]

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 
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[asterisk-users] music on hold

[Uros Djokic]

Hi,

You can convert your music files in 8000 hz and mono with sox command like
this
sox yourfile.wav -r 8000 -c 1 yournewfile,wav resample -ql

Uros

-- 
Use Free Software http://www.fsf.org/
---
Four essential software freedoms:
1) To study source code
2) To copy program
3) To modify source code
4) To redistribute modified program under condition that new user has all 4
freedoms.
Richard M. Stallman
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[asterisk-users] Music On Hold

[Idris AVCI]

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
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[asterisk-users] Music on hold

[Peter Vedstesen]

Hey.

 

Maybe this question has been asked before, but I am new here.

 

I would like to use streamed radio station as "music on hold".

 

I have tried mpg123. That doesn't work.

 

I have installed mplayer and it can connect to the stream.

> mplayer http://media06.webpartner.dk/100fm2?MSWMExt=.asf

 

But how to get mplayer and asterisk to work together?

 

My setup is trixbox 1.2.3 

 

Hoping someone know how to put asterisk and mplayer to work.

 

Regards 

Peter Vedstesen

  

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[asterisk-users] Music on hold

[Vazquez David]

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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[asterisk-users] Music on Hold

[Julien Claassen]

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


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

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[asterisk-users] Music on Hold

[Dan Saul]

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.
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[asterisk-users] Music On Hold

[Cyprus VoIP]

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.

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[asterisk-users] music on hold

[asterisk]

hi friends, 

as i am a beginner in voip, i had made a very simple dial
plan i had made two extentions n both are able to ring each other through
soft phone (X-Lite) 

 below is my dialplan


###


/ETC/EXTENSIONS.CONF 

[others]

[my-phones]
exten =>
2000,1,Dial(SIP/2000,10)
exten => 2000,2,Answer
exten =>
2000,3,MusicOnHold()

exten => 2001,1,Dial(SIP/2001,20)
exten =>
2001,2,Voicemail(2001,u)

exten => 2999,1,VoiceMailMain(${CALLERID(num)},s)


##
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. 

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[asterisk-users] music on hold

[asterisk]

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 

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[Asterisk-Users] Music on Hold

[Sahil Gupta]

Hi,
I've been trying to get music on hold going on one of our servers:
Upon dialling extension 005, it plays:
-- Executing WaitMusicOnHold("SIP/parssyd1-4dbe", "30") in new stack
-- Started music on hold, class 'default', on SIP/parssyd1-4dbe
However, no music in the background
MPG123 is intalled..
musiconhold.conf shows:
default => mp3:/var/lib/asterisk/mohmp3
The directory has?:
[EMAIL PROTECTED]:~# ls -al /var/lib/asterisk/mohmp3
total 6589
drwxr-xr-x  2 root root 160 2005-04-21 10:25 ./
drwxr-xr-x  8 root root 216 2005-02-17 22:48 ../
-rw-r--r--  1 root root 1939812 2005-04-21 10:25 fpm-calm-river.mp3
-rw-r--r--  1 root root 2582496 2005-04-21 10:25 fpm-sunshine.mp3
-rw-r--r--  1 root root 2217563 2005-04-21 10:25 fpm-world-mix.mp3
Any clues ?  Seems like it actions things but isn't playing the mp3 
files..

Regards,
Sahil Gupta
VoiceValley
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[Asterisk-Users] Music on hold

[Hall, Eric M.]

I have been working on the music on hold part for a few hours today and
I found something that just doesn't sound right.
 
If I just run asterisk via service "service asterisk start' everything
work but MOH
If I run it via asterisk -vgcd MOH works... 
 
 
Any idea what the difference is ?
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[Asterisk-Users] Music on Hold

[richard Coco]

Hi all,
 
i'm trying to configure MoH for OptiPoint400std SIP, but it doesn't work. If i try with a softclient moh works fine. Somebody experience with OptiPoint400? Is there additional settings on the Optipoint?
 
Any help would be much appreciated!!
thx.
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[Asterisk-Users] Music on hold

[Simon Brown]

I have set up an extension so I can dial it and listen to my MusicOnHold from
any handset.  This is what is in the extensions.conf:
exten => 997,1,MusicOnHold()
exten => 997,2,Hangup

After 180 seconds of playing, the call terminates.  Why does this happen?

Simon
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[Asterisk-Users] Music on Hold

[Elmar Haneke]

Hi,
I did recognice an rather strage behaviour on "Music on Hold":
Situation
Caller C does call Person A
Person A puts C on hold to ask B
MOH is (correctly) activated for C
After talking to B A does hangup to transfer C to B
In this moment MOH is activated for C for a moment
before C is transferred to B
The MOD can be seen on the asterisk console and it can be heared as a 
short buzz on Phone B.

Any Idea how to avoid this?
Elmar
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[Asterisk-Users] Music on Hold

[Mark Halverson]

Any way to use soundcard’s mic in for music on hold?

 

-Mark Halverson

 






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[Asterisk-Users] Music on hold

[asterisk]

Hello List!

I followed the instruction from
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf to get my
music working when i put someone on hold.

/etc/asterisk/musiconhold.conf
---
[classes]
default => mp3:/var/lib/asterisk/mohmp3



/etc/asterisk/zapata.conf
---
~# grep -v "^;" /etc/asterisk/zapata.conf

[trunkgroups]

[channels]
musiconhold=default
context=default
switchtype=national
signalling=fxo_ls
rxwink=300  ; Atlas seems to use long (250ms) winks

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1

immediate=no



ps ax:
---
892 pts/1Sl+0:03 /usr/sbin/asterisk -vvvgcd
899 pts/1S+ 0:01 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3


asterisk output:
---
Urgent handler
-- Started music on hold, class 'default', on CAPI[contr1/]/5
Urgent handler


So you can see that its playing the files.
But when i put someone on hold, he cant hear it at all.

Any idea?
Thx! Mario





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[Asterisk-Users] Music On Hold

[Bud Bach]

Help!  No Music on Hold.  Probably a novice
mistake but I can’t figure it out.  Here are the details:

 

CentOS 4.2

Asterisk 1.2.1 (Do I need to do something to get MOH to
build?)

Ztdummy loaded (conference works fine)

 

musiconhold.conf:

 

[default]

mode=quietmp3

directory=/var/lib/asterisk/mohmp3

 

Sip device (x-lite – also tried with an ATA) with canreinvite=no:

 

sip.conf:

 

[7211]

username=7211

secret=

host=dynamic

type=friend

context=standardphone

disallow=all

allow=gsm

allow=ulaw

allow=alaw

allow=g723.1

allow=g729

canreinvite=no

 

Extensions.conf:

 

exten => 8702,1,Answer()

exten => 8702,n,MusicOnHold(default)

exten => 8702,n,Hangup()

 

 

# asterisk -r

Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium.

Written by Mark Spencer <[EMAIL PROTECTED]>

=

Connected to Asterisk 1.2.1 currently running on ccsip (pid
= 4782)

Verbosity is at least 3

  == Spawn extension (standardphone, 8702, 2) exited
non-zero on 'SIP/7211-be01'

    -- Executing Answer("SIP/7211-cedb",
"") in new stack

    -- Executing MusicOnHold("SIP/7211-cedb",
"default") in new stack

    -- Started music on hold, class
'default', on channel 'SIP/7211-cedb'

    -- Stopped music on hold on SIP/7211-cedb

  == Spawn extension (standardphone, 8702, 2) exited
non-zero on 'SIP/7211-cedb'

 

The “Stopped music on hold” happens immediately
like it can’t find something.  Should I give up and use madplay?

 

-- Bud






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[Asterisk-Users] Music on Hold

[Edward0219]

I tried everything to get the music on hold feature to work with [EMAIL PROTECTED], to no avail.
 
Need some help to get it running.
 
Thanks,
 
Eduardo Zaldibar
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[Asterisk-Users] Music on hold

[amaury BOSSE]

Hi all,
I have some problems with music on hold.
It works with the default category but not with additional ones.
When I start Asterisk, 2 mpg123 processes are started with the default
moh but none with additional ones.
Does someone already had this problem and could help me?

Thanks,
Amaury

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[asterisk-users] music on hold

[salaheddine elharit]

i configure new music on hold like below in order to play music for outbond
calls

i want tp play a music until answer form customer

[default1]
mode=files
directory=/var/lib/asterisk/moh1

exten => 0678XX,1,Set(CALLERID(number)=520XX)
exten => 0678XX,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => 0678XX,n,Dial(Zap/g1/${EXTEN},,m(default1))
exten => 0678XX,n,Hangup()


when i put the default music i can listen without issue but when i put
another music .wav Or gsm or Mp3
there is no music  there is just the ringing
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[asterisk-users] Music on Hold

[taimur hasan]

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

  
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[asterisk-users] Music on hold

[Kris Stark]

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

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[Asterisk-Users] Music on hold quality..

[WipeOut .]

Hi,

Does any one have any pointers on improving moh quality??

Symptoms are crackling and hissing as the sound comes and goes..

I installed mpg123 this morning..

I have tried various MP3's sampled at 160k, 128k, 32k and 8k and they all sounded 
terrible... The PC is a P4 so its got plenty of processing power.. I have tried a few 
different types of classical music (Piano, Violin and full Orchestra)..

Thanks..
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[Asterisk-Users] music on hold help

[John Brown]

I've only gotten to hear Music On Hold once.  

I am running the right version of mpg123 .059r as
downloaded from mpg123.de and compiled locally

So I'm looking for any help on getting this up and
running.

I can see on the console that the SIP phone is placing
the call on hold, but there is no audio coming out 
on the other phone.

Mucho, thanks

john
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RE: [Asterisk-Users] Music on Hold

[Ashley Jones]

Ernest,

The site http://www.royaltyfreemusic.com/ seems to have a sample for
every song on their site.

Best,
-adj

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W.
Lessenger
Sent: Friday, September 12, 2003 11:08 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Music on Hold


Does anybody have a good source for hold music? I can see a number of 
companies on the web that sell royalty-free MOH, but they don't all
provide 
samples. The customer service desk has requested "calming, not sleeping,

but calming" and "this is a high-tech company, so make it 'techie'
[sic]".

Thanks,
--Ernest

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[Asterisk-Users] Music On Hold distorted

[Clif Jones]

I have searching the forums here on how to get Music On Hold working
and I have been able to get * to accept a command for MusicOnHold
and for Meetme after loading the ztdummy module.  I used the default
config for /etc/zaptel.conf since I saw no guidance on this.  My problem
now is that when I activate MusicOnHold, the sample music file sounds
very slow and distorted.  My best guess is that it is playing so slow it
sounds like Darth Vader breathing.  Anybody have any ideas on what
is going on?
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Re: [Asterisk-Users] Music on hold...

[Tilghman Lesher]

On Sunday 19 October 2003 18:19, Chris Hariga wrote:
> Hi,
>
> I need a sound card and mpg123 for music on hold??? When I call
> Digium the guys toll me "is not necessary to have a sound card". My
> music on hold doesn't work :((

Sound card is not necessary, but mpg123 is.  Please make sure that
you really have mpg123, not RedHat's mpg321 symlinked to mpg123.

-Tilghman

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Re: [Asterisk-Users] Music on hold...

[Juan J. Sierralta P.]

On Sun, 2003-10-19 at 21:39, CW_ASN wrote:
> No, you don't need a sound card.
> Do you have ztdummy loaded or zaptel device in your system?

AFAIK, MOH does no nees a zaptel device or zaptel dummy driver, just
MeetMe needs it.

-- 
Juanjo sin .sig

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Re: [Asterisk-Users] Music on hold...

[CW_ASN]

Yes, right... sorry.


- Original Message - 
From: "Juan J. Sierralta P." <[EMAIL PROTECTED]>
To: "Asterisk Users" <[EMAIL PROTECTED]>
Sent: Sunday, October 19, 2003 11:27 PM
Subject: Re: [Asterisk-Users] Music on hold...


> On Sun, 2003-10-19 at 21:39, CW_ASN wrote:
> > No, you don't need a sound card.
> > Do you have ztdummy loaded or zaptel device in your system?
> 
> AFAIK, MOH does no nees a zaptel device or zaptel dummy driver, just
> MeetMe needs it.
> 
> -- 
> Juanjo sin .sig
> 
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Re: [Asterisk-Users] Music on Hold

[CW_ASN]

MP3Player is not the way to have Music on Hold... Please do a test in this
way:

exten => 2091,1,Answer
exten => 2091,2,Wait,1
exten => 2091,3,MusicOnHold,default

And the musiconhold.conf:

;
; Music on hold class definitions
;
[classes]
default => quietmp3:/var/lib/asterisk/mohmp3
;loud => mp3:/var/lib/asterisk/mohmp3
;random => quietmp3:/var/lib/asterisk/mohmp3,-z

Hope this helps.

Gus

- Original Message -
From: "Phillip Jackson, Director of IT" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, October 26, 2003 4:14 AM
Subject: [Asterisk-Users] Music on Hold


> Having a weird issue with on hold music ... I do have mpg123 installed.
>
> When requesting extension  for testing, which is setup as:
>
> exten => ,1,Answer  ; Answer the line
> exten => ,2,DigitTimeout,5  ; Set Digit Timeout to 5 seconds
> exten => ,3,MP3Player(${MP3ROOT}/sample-hold.mp3)
>
> I recieve this err:
>
> -- Executing MP3Player("SIP/100-26af", "/sample-hold.mp3") in new stack
> WARNING[1217602880]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error:
> Resource temporarily unavailable
> NOTICE[1217602880]: File app_mp3.c, Line 80 (timed_read): Selected timed
> out/errored out with 0
>
> Not sure what's up...
>
> Phillip
>
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Re: [Asterisk-Users] Music on Hold

[Kang . ChenJi]

I would appreciate it if anyone can give me some instructions on how to
install mpg123.

Thanks in advance,
Kang



   
 
  "Phillip Jackson, Director   
 
  of IT"To:   <[EMAIL PROTECTED]>  
   
  <[EMAIL PROTECTED]> cc:  
   
  Sent by:      Subject:  [Asterisk-Users] 
Music on Hold
  [EMAIL PROTECTED]

  .digium.com  
 
   
 
   
 
  10/26/2003 02:14 AM  
 
  Please respond to
 
  asterisk-users   
 
   
 
   
 




Having a weird issue with on hold music ... I do have mpg123 installed.

When requesting extension  for testing, which is setup as:

exten => ,1,Answer  ; Answer the line
exten => ,2,DigitTimeout,5  ; Set Digit Timeout to 5 seconds
exten => ,3,MP3Player(${MP3ROOT}/sample-hold.mp3)

I recieve this err:

-- Executing MP3Player("SIP/100-26af", "/sample-hold.mp3") in new stack
WARNING[1217602880]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error:
Resource temporarily unavailable
NOTICE[1217602880]: File app_mp3.c, Line 80 (timed_read): Selected timed
out/errored out with 0

Not sure what's up...

Phillip

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Re: [Asterisk-Users] Music on Hold

[Eric Wieling]

The mpg123 homepage is at http://www.mpg123.de/  Either follow the
instructions there for downloading and building mpg123 or use whatever
installation tool your Linux distro uses.

On Mon, 2003-10-27 at 14:25, [EMAIL PROTECTED]
wrote:
> I would appreciate it if anyone can give me some instructions on how to
> install mpg123.
> 
> Thanks in advance,
> Kang
> 
> 
> 
>  
>
>   "Phillip Jackson, Director 
>
>   of IT"To:   <[EMAIL 
> PROTECTED]> 
>   <[EMAIL PROTECTED]> cc:
>  
>           Sent by:  Subject:  [Asterisk-Users] 
> Music on Hold
>   [EMAIL PROTECTED]  
>   
>   .digium.com
>
>  
>
>  
>
>   10/26/2003 02:14 AM
>
>   Please respond to  
>
>   asterisk-users 
>
>  
>
>  
>
> 
> 
> 
> 
> Having a weird issue with on hold music ... I do have mpg123 installed.
> 
> When requesting extension  for testing, which is setup as:
> 
> exten => ,1,Answer  ; Answer the line
> exten => ,2,DigitTimeout,5  ; Set Digit Timeout to 5 seconds
> exten => ,3,MP3Player(${MP3ROOT}/sample-hold.mp3)
> 
> I recieve this err:
> 
> -- Executing MP3Player("SIP/100-26af", "/sample-hold.mp3") in new stack
> WARNING[1217602880]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error:
> Resource temporarily unavailable
> NOTICE[1217602880]: File app_mp3.c, Line 80 (timed_read): Selected timed
> out/errored out with 0
> 
> Not sure what's up...
> 
> Phillip
> 
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> 
> 
> 
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RE: [Asterisk-Users] Music on Hold

[Ray Burkholder]

Some notes can be found at 
http://www.oneunified.net/support/asterisk/index.html


Regards,
Ray Burkholder



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> [EMAIL PROTECTED]
> Sent: October 27, 2003 15:25
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Music on Hold
> 
> 
> 
> I would appreciate it if anyone can give me some instructions 
> on how to
> install mpg123.
> 
> Thanks in advance,
> Kang
> 
> 
> 
>   
>   
> 
>   "Phillip Jackson, Director  
>   
> 
>   of IT"To:   
> <[EMAIL PROTECTED]> 
> 
>   <[EMAIL PROTECTED]> cc:   
>       
>     
>   Sent by:  
> Subject:  [Asterisk-Users] Music on Hold  
>   
>   [EMAIL PROTECTED]  
>   
> 
>   .digium.com 
>   
> 
>   
>   
> 
>   
>   
> 
>   10/26/2003 02:14 AM 
>   
> 
>   Please respond to   
>   
> 
>   asterisk-users  
>   
> 
>   
>   
> 
>   
>   
> 
> 
> 
> 
> 
> Having a weird issue with on hold music ... I do have mpg123 
> installed.
> 
> When requesting extension  for testing, which is setup as:
> 
> exten => ,1,Answer  ; Answer the line
> exten => ,2,DigitTimeout,5  ; Set Digit Timeout 
> to 5 seconds
> exten => ,3,MP3Player(${MP3ROOT}/sample-hold.mp3)
> 
> I recieve this err:
> 
> -- Executing MP3Player("SIP/100-26af", "/sample-hold.mp3") in 
> new stack
> WARNING[1217602880]: File rtp.c, Line 374 (ast_rtp_read): RTP 
> Read error:
> Resource temporarily unavailable
> NOTICE[1217602880]: File app_mp3.c, Line 80 (timed_read): 
> Selected timed
> out/errored out with 0
> 
> Not sure what's up...
> 
> Phillip
> 
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Re: [Asterisk-Users] Music on Hold

[Rich Adamson]

> I would appreciate it if anyone can give me some instructions on how to
> install mpg123.

One more time for those running RedHat v9 ... ;)



Well... we found the problem. Redhat guys replaced mpg123 for mpg321.
Asterisk only works with original mpg123.

First, shutdown asterisk and kill all mpg process:

killall -9 mpg123

Second, remove the symbolic links mpg123 located in /usr/bin and
/usr/local/bin:
rm /usr/bin/mpg123
rm /usr/local/bin/mpg123(if exists)

Third, you need to download
http://www.mpg123.de/mpg123/precompiled/mpg123-0.59q-1.i386.rpm
And install using rpm -ivh mpg123-0.59q-1.i386.rpm

Fourth step is:
cp /usr/local/bin/mpg123 /usr/bin/mpg123

Last step (obvioulsy) is start asterisk again.

Thats it. I hope this works.



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[Asterisk-Users] Music on Hold Warnings

[Craig Waddington]

Hi.

 

I am having the following warning when using music on hold.

 

It works from X-Lite to Grandstream. I get a lot of errors
and warnings.

 

1.Warning, flexibel rate not heavily tested!

 

2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread:
Request to schedule in the past?!?!

 

Thanks for any help.

 

 

Full Output below:

 

Jan 30 10:24:55 WARNING[1133718080]: chan_sip.c:486
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 909 (Response)

Jan 30 10:24:55 WARNING[1217602880]: file.c:521 ast_readaudio_callback:
Failed to write frame

  == Spawn extension (sip, 5001, 2) exited non-zero on
'SIP/5002-0922'

    -- SIP/5001-6a4d answered SIP/5002-d365

    -- Attempting native bridge of SIP/5002-d365 and
SIP/5001-6a4d

    -- Started music on hold, class 'default', on
SIP/5001-6a4d

Warning, flexibel rate not heavily tested!

Jan 30 10:25:14 NOTICE[1100258240]: res_musiconhold.c:260
monmp3thread: Request to schedule in the past?!?!

    -- Stopped music on hold on SIP/5001-6a4d

  == Spawn extension (sip, 5001, 1) exited non-zero on
'SIP/5002-d365'

 

    -- Executing Dial("SIP/5002-a28b",
"SIP/5001|20") in new stack

    -- Called 5001

    -- SIP/5001-87f7 is ringing

    -- SIP/5001-87f7 answered SIP/5002-a28b

    -- Attempting native bridge of SIP/5002-a28b and SIP/5001-87f7

    -- Started music on hold, class 'default', on
SIP/5002-a28b

Warning, flexibel rate not heavily tested!

Jan 30 10:26:40 NOTICE[1100258240]: res_musiconhold.c:260
monmp3thread: Request to schedule in the past?!?!

Jan 30 10:26:50 NOTICE[1234379840]: rtp.c:264
process_rfc3389: RFC3389 support incomplete.  Turn off on client if possible

    -- Stopped music on hold on SIP/5002-a28b

  == Spawn extension (sip, 5001, 1) exited non-zero on
'SIP/5002-a28b'

[Asterisk-Users] Music on Hold - Context

[AstGrp]

I have set up a * box supporting 3 different companies but have some
questions regarding MOH.  Can MOH support multiple context or classes.
Reason I ask each company would like to have different MOH sound files.
Is this possible? 

Thanks,

-gcc

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Re: [asterisk-users] Music on hold

[Kevin P. Fleming]

Fons van der Beek wrote:
> Because i want a ringing signal while people are in a waiting queue i've 
> created a wav file containing our local ringing indication
> If I make an inside call to the queue, the correct sound is played, but 
> when i make an external call, no signal is heard.
> everything else looks ok, and all other functions are ok

The Queue() application has an option to generate ringback to callers
instead of music on hold, why don't you just use that instead of trying
to craft a new solution?

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] Music on hold

[Fons van der Beek]

I tried that, its gives me the same problem.

Kevin P. Fleming schreef:
> Fons van der Beek wrote:
>   
>> Because i want a ringing signal while people are in a waiting queue i've 
>> created a wav file containing our local ringing indication
>> If I make an inside call to the queue, the correct sound is played, but 
>> when i make an external call, no signal is heard.
>> everything else looks ok, and all other functions are ok
>> 
>
> The Queue() application has an option to generate ringback to callers
> instead of music on hold, why don't you just use that instead of trying
> to craft a new solution?
>
>   


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Re: [asterisk-users] Music on hold

[Eric Wieling]

This problem would happen if you did not have /etc/asterisk/indications.conf

Fons van der Beek wrote:
> I tried that, its gives me the same problem.
> 
> Kevin P. Fleming schreef:
>> Fons van der Beek wrote:
>>   
>>> Because i want a ringing signal while people are in a waiting queue i've 
>>> created a wav file containing our local ringing indication
>>> If I make an inside call to the queue, the correct sound is played, but 
>>> when i make an external call, no signal is heard.
>>> everything else looks ok, and all other functions are ok
>>> 
>> The Queue() application has an option to generate ringback to callers
>> instead of music on hold, why don't you just use that instead of trying
>> to craft a new solution?
>>
>>   
> 
> 
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Re: [asterisk-users] Music on hold

[Fons van der Beek]

Tnx.

I checked /etc/asterisk/indications.conf and my default location nl
is listed in the options

So i am still puzzled


my extensions.conf in respect to incomming calls (as basic as possible)

exten => s,1,Answer
exten => s,2,queue(receptie|r)
exten => s,3,Voicemail(201)

everything else works as it should work, but no "ringing" on an external 
line


on the other hand, internaly: it's ok

exten => 205,1,queue(receptie|r)
exten => 205,2,busy

205 gives ringing


Eric Wieling schreef:

This problem would happen if you did not have /etc/asterisk/indications.conf

Fons van der Beek wrote:
  

I tried that, its gives me the same problem.

Kevin P. Fleming schreef:


Fons van der Beek wrote:
  
  
Because i want a ringing signal while people are in a waiting queue i've 
created a wav file containing our local ringing indication
If I make an inside call to the queue, the correct sound is played, but 
when i make an external call, no signal is heard.

everything else looks ok, and all other functions are ok



The Queue() application has an option to generate ringback to callers
instead of music on hold, why don't you just use that instead of trying
to craft a new solution?

  
  

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Re: [asterisk-users] Music on hold

[Eric Wieling]

Don't answer the line.  Also try using the US indications, just in case 
something odd is in the NL setup.

Fons van der Beek wrote:
> Tnx.
> 
> I checked /etc/asterisk/indications.conf and my default location nl
> is listed in the options
> 
> So i am still puzzled
> 
> 
> my extensions.conf in respect to incomming calls (as basic as possible)
> 
> exten => s,1,Answer
> exten => s,2,queue(receptie|r)
> exten => s,3,Voicemail(201)
> 
> everything else works as it should work, but no "ringing" on an external 
> line
> 
> on the other hand, internaly: it's ok
> 
> exten => 205,1,queue(receptie|r)
> exten => 205,2,busy
> 
> 205 gives ringing
> 
> 
> Eric Wieling schreef:
>> This problem would happen if you did not have 
>> /etc/asterisk/indications.conf
>>
>> Fons van der Beek wrote:
>>  
>>> I tried that, its gives me the same problem.
>>>
>>> Kevin P. Fleming schreef:
>>>
 Fons van der Beek wrote:

> Because i want a ringing signal while people are in a waiting queue 
> i've created a wav file containing our local ringing indication
> If I make an inside call to the queue, the correct sound is played, 
> but when i make an external call, no signal is heard.
> everything else looks ok, and all other functions are ok
> 
 The Queue() application has an option to generate ringback to callers
 instead of music on hold, why don't you just use that instead of trying
 to craft a new solution?

 
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>>>
>>> 
>>
>>   
> 
> 
> 
> 
> 
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T-1, PRI, Frame Relay, Linux, and network design.  Based near 
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Re: [asterisk-users] Music on hold

[Eric Wieling]

Replying to my own post.  Asterisk uses indications.conf when it has to 
provide tones AFTER the line is answered.  You might get a message on 
the console like "Unable to handle indication 15" or something like that.

Eric Wieling wrote:
> Don't answer the line.  Also try using the US indications, just in case 
> something odd is in the NL setup.
> 
> Fons van der Beek wrote:
>> Tnx.
>>
>> I checked /etc/asterisk/indications.conf and my default location nl
>> is listed in the options
>>
>> So i am still puzzled
>>
>>
>> my extensions.conf in respect to incomming calls (as basic as possible)
>>
>> exten => s,1,Answer
>> exten => s,2,queue(receptie|r)
>> exten => s,3,Voicemail(201)
>>
>> everything else works as it should work, but no "ringing" on an external 
>> line
>>
>> on the other hand, internaly: it's ok
>>
>> exten => 205,1,queue(receptie|r)
>> exten => 205,2,busy
>>
>> 205 gives ringing
>>
>>
>> Eric Wieling schreef:
>>> This problem would happen if you did not have 
>>> /etc/asterisk/indications.conf
>>>
>>> Fons van der Beek wrote:
>>>  
 I tried that, its gives me the same problem.

 Kevin P. Fleming schreef:

> Fons van der Beek wrote:
>
>> Because i want a ringing signal while people are in a waiting queue 
>> i've created a wav file containing our local ringing indication
>> If I make an inside call to the queue, the correct sound is played, 
>> but when i make an external call, no signal is heard.
>> everything else looks ok, and all other functions are ok
>> 
> The Queue() application has an option to generate ringback to callers
> instead of music on hold, why don't you just use that instead of trying
> to craft a new solution?
>
> 
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>>>   
>>
>>
>> 
>>
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Re: [asterisk-users] Music on hold

[Fons van der Beek]

it's very odd
-I just upgraded to 1.4.18 (from 1.4.17)
-removed answer
-changed to several other options, still no luck
(restarted also)





Eric Wieling schreef:
Don't answer the line.  Also try using the US indications, just in case 
something odd is in the NL setup.


Fons van der Beek wrote:
  

Tnx.

I checked /etc/asterisk/indications.conf and my default location nl
is listed in the options

So i am still puzzled


my extensions.conf in respect to incomming calls (as basic as possible)

exten => s,1,Answer
exten => s,2,queue(receptie|r)
exten => s,3,Voicemail(201)

everything else works as it should work, but no "ringing" on an external 
line


on the other hand, internaly: it's ok

exten => 205,1,queue(receptie|r)
exten => 205,2,busy

205 gives ringing


Eric Wieling schreef:

This problem would happen if you did not have 
/etc/asterisk/indications.conf


Fons van der Beek wrote:
 
  

I tried that, its gives me the same problem.

Kevin P. Fleming schreef:
   


Fons van der Beek wrote:
   
  
Because i want a ringing signal while people are in a waiting queue 
i've created a wav file containing our local ringing indication
If I make an inside call to the queue, the correct sound is played, 
but when i make an external call, no signal is heard.

everything else looks ok, and all other functions are ok



The Queue() application has an option to generate ringback to callers
instead of music on hold, why don't you just use that instead of trying
to craft a new solution?


  

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Re: [asterisk-users] Music on hold

[Fons van der Beek]

NOT answering did the trick!
Tnx a lot! now it works like it should work!



Eric Wieling schreef:
Replying to my own post.  Asterisk uses indications.conf when it has to 
provide tones AFTER the line is answered.  You might get a message on 
the console like "Unable to handle indication 15" or something like that.


Eric Wieling wrote:
  
Don't answer the line.  Also try using the US indications, just in case 
something odd is in the NL setup.


Fons van der Beek wrote:


Tnx.

I checked /etc/asterisk/indications.conf and my default location nl
is listed in the options

So i am still puzzled


my extensions.conf in respect to incomming calls (as basic as possible)

exten => s,1,Answer
exten => s,2,queue(receptie|r)
exten => s,3,Voicemail(201)

everything else works as it should work, but no "ringing" on an external 
line


on the other hand, internaly: it's ok

exten => 205,1,queue(receptie|r)
exten => 205,2,busy

205 gives ringing


Eric Wieling schreef:
  
This problem would happen if you did not have 
/etc/asterisk/indications.conf


Fons van der Beek wrote:
 


I tried that, its gives me the same problem.

Kevin P. Fleming schreef:
   
  

Fons van der Beek wrote:
   

Because i want a ringing signal while people are in a waiting queue 
i've created a wav file containing our local ringing indication
If I make an inside call to the queue, the correct sound is played, 
but when i make an external call, no signal is heard.

everything else looks ok, and all other functions are ok

  

The Queue() application has an option to generate ringback to callers
instead of music on hold, why don't you just use that instead of trying
to craft a new solution?




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Re: [asterisk-users] Music on hold

[Eric Wieling]

If not answering fixes the problem then the issue is indications.conf. 
Try using the indications.conf.sample file included with the Asterisk 
source code, then stop Asterisk and starting it again.  I do not know if 
indications.conf is reloaded on a reload.

Fons van der Beek wrote:
> NOT answering did the trick!
> Tnx a lot! now it works like it should work!
> 
> 
> 
> Eric Wieling schreef:
>> Replying to my own post.  Asterisk uses indications.conf when it has 
>> to provide tones AFTER the line is answered.  You might get a message 
>> on the console like "Unable to handle indication 15" or something like 
>> that.
>>
>> Eric Wieling wrote:
>>  
>>> Don't answer the line.  Also try using the US indications, just in 
>>> case something odd is in the NL setup.
>>>
>>> Fons van der Beek wrote:
>>>
 Tnx.

 I checked /etc/asterisk/indications.conf and my default location nl
 is listed in the options

 So i am still puzzled


 my extensions.conf in respect to incomming calls (as basic as possible)

 exten => s,1,Answer
 exten => s,2,queue(receptie|r)
 exten => s,3,Voicemail(201)

 everything else works as it should work, but no "ringing" on an 
 external line

 on the other hand, internaly: it's ok

 exten => 205,1,queue(receptie|r)
 exten => 205,2,busy

 205 gives ringing


 Eric Wieling schreef:
  
> This problem would happen if you did not have 
> /etc/asterisk/indications.conf
>
> Fons van der Beek wrote:
>  
>
>> I tried that, its gives me the same problem.
>>
>> Kevin P. Fleming schreef:
>> 
>>> Fons van der Beek wrote:
>>>   
 Because i want a ringing signal while people are in a waiting 
 queue i've created a wav file containing our local ringing 
 indication
 If I make an inside call to the queue, the correct sound is 
 played, but when i make an external call, no signal is heard.
 everything else looks ok, and all other functions are ok
   
>>> The Queue() application has an option to generate ringback to 
>>> callers
>>> instead of music on hold, why don't you just use that instead of 
>>> trying
>>> to craft a new solution?
>>>
>>> 
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>>
>>   
>   
  


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>>
>>   
> 
> 
> 
> 
> 
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Re: [asterisk-users] Music on hold

[Fons van der Beek]

I've overwritten the indications.conf with the one from the sourcecode, 
stil no luck
Perhaps somebody knows what the correct value for indications.conf is 
when using the dutch xs4all as sip carrier??


and even with verbose set to 114 (quite big) there are no errormessages 
indicating that something is wrong with indications (in respect to syntax)


Eric Wieling schreef:
If not answering fixes the problem then the issue is indications.conf. 
Try using the indications.conf.sample file included with the Asterisk 
source code, then stop Asterisk and starting it again.  I do not know if 
indications.conf is reloaded on a reload.


Fons van der Beek wrote:
  

NOT answering did the trick!
Tnx a lot! now it works like it should work!



Eric Wieling schreef:

Replying to my own post.  Asterisk uses indications.conf when it has 
to provide tones AFTER the line is answered.  You might get a message 
on the console like "Unable to handle indication 15" or something like 
that.


Eric Wieling wrote:
 
  
Don't answer the line.  Also try using the US indications, just in 
case something odd is in the NL setup.


Fons van der Beek wrote:
   


Tnx.

I checked /etc/asterisk/indications.conf and my default location nl
is listed in the options

So i am still puzzled


my extensions.conf in respect to incomming calls (as basic as possible)

exten => s,1,Answer
exten => s,2,queue(receptie|r)
exten => s,3,Voicemail(201)

everything else works as it should work, but no "ringing" on an 
external line


on the other hand, internaly: it's ok

exten => 205,1,queue(receptie|r)
exten => 205,2,busy

205 gives ringing


Eric Wieling schreef:
 
  
This problem would happen if you did not have 
/etc/asterisk/indications.conf


Fons van der Beek wrote:
 
   


I tried that, its gives me the same problem.

Kevin P. Fleming schreef:

  

Fons van der Beek wrote:
  

Because i want a ringing signal while people are in a waiting 
queue i've created a wav file containing our local ringing 
indication
If I make an inside call to the queue, the correct sound is 
played, but when i make an external call, no signal is heard.

everything else looks ok, and all other functions are ok
  
  
The Queue() application has an option to generate ringback to 
callers
instead of music on hold, why don't you just use that instead of 
trying

to craft a new solution?




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Re: [asterisk-users] Music on hold

[Trevor Peirce]

Fons van der Beek wrote:
> I've overwritten the indications.conf with the one from the 
> sourcecode, stil no luck
> Perhaps somebody knows what the correct value for indications.conf is 
> when using the dutch xs4all as sip carrier??

A simple way for you to test your indications.conf as far as the ringing 
goes is something like this:

exten => s,1,Answer
exten => s,n,PlayTones(ring)
exten => s,n,Wait(30)
exten => s,n,Hangup

That should pick up the line and then play your locale's ring tone for 
30 seconds before hanging up.  If you hear ringing then indications.conf 
is fine, otherwise you have confirmed that there is a problem somewhere.

This will have nothing to do with your carrier as the sounds are 
generated by asterisk itself as audio (as opposed to any kind of 
carrier-specific signaling).

Trevor

Real CNAM data for incoming Caller ID @ www.cnam.info



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Re: [asterisk-users] Music on hold

[Fons van der Beek]

I guess we are back to the fundamental problem: "no asterisk generated 
sounds on the external call"


After implementing the described test for indications.conf
The CLI outputted:
-- Executing [EMAIL PROTECTED]:1] Answer("SIP/0475769XXX-095a8488", "") in new 
stack
   -- Executing [EMAIL PROTECTED]:2] PlayTones("SIP/0475769XXX-095a8488", 
"ring") in new stack
   -- Executing [EMAIL PROTECTED]:3] Wait("SIP/0475769XXX-095a8488", "30") in 
new stack


This looks OK, but there is no sound to be heard on the other end.

Sip show peers for the other end shows:
* Name   : sip.xs4all.nl
 Secret   : 
 MD5Secret: 
 Context  : default
 Subscr.Cont. : default
 Language : en
 AMA flags: Unknown
 Transfer mode: open
 CallingPres  : Presentation Allowed, Not Screened
 FromUser : 0475769XXX
 FromDomain   : sip.xs4all.nl
 Callgroup:
 Pickupgroup  :
 Mailbox  :
 VM Extension : asterisk
 LastMsgsSent : 32767/65535
 Call limit   : 0
 Dynamic  : No
 Callerid : "" <>
 MaxCallBR: 384 kbps
 Expire   : -1
 Insecure : port,invite
 Nat  : RFC3581
 ACL  : No
 T38 pt UDPTL : No
 CanReinvite  : No
 PromiscRedir : No
 User=Phone   : No
 Video Support: Yes
 Trust RPID   : No
 Send RPID: No
 Subscriptions: Yes
 Overlap dial : No
 DTMFmode : auto
 LastMsg  : 0
 ToHost   : sip.xs4all.nl
 Addr->IP : 82.101.XX.XX Port 5060
 Defaddr->IP  : 0.0.0.0 Port 0
 Def. Username: 0475769XXX
 SIP Options  : (none)
 Codecs   : 0x104 (ulaw|g729)
 Codec Order  : (ulaw:20,g729:20)
 Auto-Framing:  No
 Status   : Unmonitored
 Useragent:
 Reg. Contact :








Trevor Peirce schreef:

Fons van der Beek wrote:
  
I've overwritten the indications.conf with the one from the 
sourcecode, stil no luck
Perhaps somebody knows what the correct value for indications.conf is 
when using the dutch xs4all as sip carrier??



A simple way for you to test your indications.conf as far as the ringing 
goes is something like this:


exten => s,1,Answer
exten => s,n,PlayTones(ring)
exten => s,n,Wait(30)
exten => s,n,Hangup

That should pick up the line and then play your locale's ring tone for 
30 seconds before hanging up.  If you hear ringing then indications.conf 
is fine, otherwise you have confirmed that there is a problem somewhere.


This will have nothing to do with your carrier as the sounds are 
generated by asterisk itself as audio (as opposed to any kind of 
carrier-specific signaling).


Trevor

Real CNAM data for incoming Caller ID @ www.cnam.info



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Re: [asterisk-users] Music on hold

[Fons van der Beek]

While the call is progressing

sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Format   
Hold Last Message
82.101.62.XX 0475769XXX  14151-EX-29  00101/703757593  0x4 
(ulaw)   No   Rx: ACK

82.101.62.XX 0475769XXX  6ec6f62d57d  00103/0  0x0 (nothing)No

Codec=Ulaw, still no "ringing"

Fons van der Beek schreef:
I guess we are back to the fundamental problem: "no asterisk generated 
sounds on the external call"


After implementing the described test for indications.conf
The CLI outputted:
 -- Executing [EMAIL PROTECTED]:1] Answer("SIP/0475769XXX-095a8488", "") in 
new stack
-- Executing [EMAIL PROTECTED]:2] PlayTones("SIP/0475769XXX-095a8488", 
"ring") in new stack
-- Executing [EMAIL PROTECTED]:3] Wait("SIP/0475769XXX-095a8488", "30") 
in new stack


This looks OK, but there is no sound to be heard on the other end.

Sip show peers for the other end shows:
* Name   : sip.xs4all.nl
  Secret   : 
  MD5Secret: 
  Context  : default
  Subscr.Cont. : default
  Language : en
  AMA flags: Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  FromUser : 0475769XXX
  FromDomain   : sip.xs4all.nl
  Callgroup:
  Pickupgroup  :
  Mailbox  :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic  : No
  Callerid : "" <>
  MaxCallBR: 384 kbps
  Expire   : -1
  Insecure : port,invite
  Nat  : RFC3581
  ACL  : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: Yes
  Trust RPID   : No
  Send RPID: No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode : auto
  LastMsg  : 0
  ToHost   : sip.xs4all.nl
  Addr->IP : 82.101.XX.XX Port 5060
  Defaddr->IP  : 0.0.0.0 Port 0
  Def. Username: 0475769XXX
  SIP Options  : (none)
  Codecs   : 0x104 (ulaw|g729)
  Codec Order  : (ulaw:20,g729:20)
  Auto-Framing:  No
  Status   : Unmonitored
  Useragent:
  Reg. Contact :








Trevor Peirce schreef:

Fons van der Beek wrote:
  
I've overwritten the indications.conf with the one from the 
sourcecode, stil no luck
Perhaps somebody knows what the correct value for indications.conf is 
when using the dutch xs4all as sip carrier??



A simple way for you to test your indications.conf as far as the ringing 
goes is something like this:


exten => s,1,Answer
exten => s,n,PlayTones(ring)
exten => s,n,Wait(30)
exten => s,n,Hangup

That should pick up the line and then play your locale's ring tone for 
30 seconds before hanging up.  If you hear ringing then indications.conf 
is fine, otherwise you have confirmed that there is a problem somewhere.


This will have nothing to do with your carrier as the sounds are 
generated by asterisk itself as audio (as opposed to any kind of 
carrier-specific signaling).


Trevor

Real CNAM data for incoming Caller ID @ www.cnam.info



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Re: [asterisk-users] Music on hold

[Trevor Peirce]

Fons van der Beek wrote:
> After implementing the described test for indications.conf
> The CLI outputted:
>  -- Executing [EMAIL PROTECTED]:1] Answer("SIP/0475769XXX-095a8488", "") in 
> new stack
> -- Executing [EMAIL PROTECTED]:2] PlayTones("SIP/0475769XXX-095a8488", 
> "ring") in new stack
> -- Executing [EMAIL PROTECTED]:3] Wait("SIP/0475769XXX-095a8488", "30") 
> in new stack
>
> This looks OK, but there is no sound to be heard on the other end.

Alright, well let's see what "ring" actually is set to for your system.

Let's see this from the command line:

cat /etc/asterisk/indications.conf | grep country=

And this from asterisk:

show indications XX  (where XX is your locale, of course).

-- 
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Re: [asterisk-users] Music on hold

[Fons van der Beek]

Tnx for your support Trevor!!

cat /etc/asterisk/indications.conf | grep country=
country=nl  ; default location


show indications nl
Country Indication  PlayList
=
nl 1000,4000
nl  dial425
nl  busy425/500,0/500
nl  ring425/1000,0/4000
nl  congestion  425/250,0/250
nl  callwaiting 425/500,0/9500
nl  dialrecall  425/500,0/50
nl  record  1400/500,0/15000
nl  info950/330,1400/330,1800/330,0/1000
nl  stutter 425/500,0/50
The 'show indications' command is deprecated and will be removed in a 
future release. Please use 'indication show' instead.



But Trevor, I guess this isn't the problem, because when i call from an 
internal location
the indication is all  right

Also moh works from internal SIP phones to the queue.
I only have a problem when i call into my asterisk box from the outside.



Trevor Peirce schreef:
> Fons van der Beek wrote:
>   
>> After implementing the described test for indications.conf
>> The CLI outputted:
>>  -- Executing [EMAIL PROTECTED]:1] Answer("SIP/0475769XXX-095a8488", "") in 
>> new stack
>> -- Executing [EMAIL PROTECTED]:2] PlayTones("SIP/0475769XXX-095a8488", 
>> "ring") in new stack
>> -- Executing [EMAIL PROTECTED]:3] Wait("SIP/0475769XXX-095a8488", "30") 
>> in new stack
>>
>> This looks OK, but there is no sound to be heard on the other end.
>> 
>
> Alright, well let's see what "ring" actually is set to for your system.
>
> Let's see this from the command line:
>
> cat /etc/asterisk/indications.conf | grep country=
>
> And this from asterisk:
>
> show indications XX  (where XX is your locale, of course).
>
>   


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Re: [asterisk-users] Music on hold

[Jared Smith]

On Sat, 2008-02-23 at 10:28 +0100, Fons van der Beek wrote:
> I guess we are back to the fundamental problem: "no asterisk generated
> sounds on the external call"

Do you have any T1/E1 cards in your system that aren't configured?  If a
zaptel card isn't taking interrupts, that would cause this same type of
problem.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Music on hold

[Fons van der Beek]

Jared YES
That seems to be the problem!

A very very long time ago I installed a X101P (an original one) and 
forgot about it.


After issuing a modprobe ztdummy, indications on the outside line 
indication work as they should.

After that i configured my X101P the way it should be configured!

And yes! now indications are the way they should be
I rebooted, I restarted asterisk and it keeps working!


I want to thank everyone who helped me, Thank you all



Jared Smith schreef:

On Sat, 2008-02-23 at 10:28 +0100, Fons van der Beek wrote:
  

I guess we are back to the fundamental problem: "no asterisk generated
sounds on the external call"



Do you have any T1/E1 cards in your system that aren't configured?  If a
zaptel card isn't taking interrupts, that would cause this same type of
problem.

  


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Re: [asterisk-users] Music on hold

[Eric Wieling]

I must have started reading this thread after you reported that you 
actually had an AUDIO problem rather than a RINGBACK problem.

The issue you experienced is a common one.  Someday I hope Digium fixes 
that bug/design flaw.

Fons van der Beek wrote:
> Jared YES
> That seems to be the problem!
> 
> A very very long time ago I installed a X101P (an original one) and 
> forgot about it.
> 
> After issuing a modprobe ztdummy, indications on the outside line 
> indication work as they should.
> After that i configured my X101P the way it should be configured!
> 
> And yes! now indications are the way they should be
> I rebooted, I restarted asterisk and it keeps working!
> 
> 
> I want to thank everyone who helped me, Thank you all
> 
> 
> 
> Jared Smith schreef:
>> On Sat, 2008-02-23 at 10:28 +0100, Fons van der Beek wrote:
>>  
>>> I guess we are back to the fundamental problem: "no asterisk generated
>>> sounds on the external call"
>>> 
>>
>> Do you have any T1/E1 cards in your system that aren't configured?  If a
>> zaptel card isn't taking interrupts, that would cause this same type of
>> problem.
>>
>>   
> 
> 
> 
> 
> 
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Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
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Re: [asterisk-users] Music on hold

[Fons van der Beek]

but your support was superior Eric!
tnx for your help!


Eric Wieling schreef:
I must have started reading this thread after you reported that you 
actually had an AUDIO problem rather than a RINGBACK problem.


The issue you experienced is a common one.  Someday I hope Digium fixes 
that bug/design flaw.


Fons van der Beek wrote:
  

Jared YES
That seems to be the problem!

A very very long time ago I installed a X101P (an original one) and 
forgot about it.


After issuing a modprobe ztdummy, indications on the outside line 
indication work as they should.

After that i configured my X101P the way it should be configured!

And yes! now indications are the way they should be
I rebooted, I restarted asterisk and it keeps working!


I want to thank everyone who helped me, Thank you all



Jared Smith schreef:


On Sat, 2008-02-23 at 10:28 +0100, Fons van der Beek wrote:
 
  

I guess we are back to the fundamental problem: "no asterisk generated
sounds on the external call"



Do you have any T1/E1 cards in your system that aren't configured?  If a
zaptel card isn't taking interrupts, that would cause this same type of
problem.

  
  




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Re: [asterisk-users] Music On Hold

[Forrest Beck]

Make the file the only one in the /var/lib/asterisk/moh directory.

Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz



On Sep 26, 2007, at 3:07 AM, Joel Hill wrote:


Hi All,

I need to have the same file played from MoH every time someone  
gets to
MoH from a Dial. I want to play marketing messages from it and I  
want it

to start from file 1 every time.

Anyone know if/how this can be done?

Cheers,

Joel.


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Re: [asterisk-users] Music On Hold

[Mojo with Horan & Company, LLC]

So concatenate all the files you've got into one to follow Forrest's 
suggestion :)


Forrest Beck wrote:
> Make the file the only one in the /var/lib/asterisk/moh directory.
>
> Forrest Beck
> [EMAIL PROTECTED] 
> www.shift8.biz
>
>
>
> On Sep 26, 2007, at 3:07 AM, Joel Hill wrote:
>
>> Hi All,
>>
>> I need to have the same file played from MoH every time someone gets to
>> MoH from a Dial. I want to play marketing messages from it and I want it
>> to start from file 1 every time.
>>
>> Anyone know if/how this can be done?
>>
>> Cheers,
>>
>> Joel.
>>
>>
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Re: [asterisk-users] Music On Hold

[Joel Hill]

Thanks for the suggestion, but I need it to play multiple messages.
Always starting with the same one.

Cheers,

Joel.

On Wed, 2007-09-26 at 09:36 -0400, Forrest Beck wrote:
> Make the file the only one in the /var/lib/asterisk/moh directory.
> 
> Forrest Beck
> [EMAIL PROTECTED]
> www.shift8.biz
> 
> 
> 
> 
> 
> On Sep 26, 2007, at 3:07 AM, Joel Hill wrote:
> 
> > Hi All,
> > 
> > 
> > I need to have the same file played from MoH every time someone gets
> > to
> > MoH from a Dial. I want to play marketing messages from it and I
> > want it
> > to start from file 1 every time.
> > 
> > 
> > Anyone know if/how this can be done?
> > 
> > 
> > Cheers,
> > 
> > 
> > Joel.
> > 
> > 
> > 
> > 
> > ___
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Re: [asterisk-users] Music On Hold

[David Gomillion]

> > Hi All,
> >
> > I need to have the same file played from MoH every time someone gets
> > to
> > MoH from a Dial. I want to play marketing messages from it and I
> > want it
> > to start from file 1 every time.
> >
> > Anyone know if/how this can be done?

> On Wed, 2007-09-26 at 09:36 -0400, Forrest Beck wrote:
> > Make the file the only one in the /var/lib/asterisk/moh directory.
> >
> > Forrest Beck
> > [EMAIL PROTECTED]
> > www.shift8.biz
> Thanks for the suggestion, but I need it to play multiple messages.
> Always starting with the same one.
>
> Cheers,
>
> Joel.


Create a new MOH class with one large file consisting of every message you
want heard, in the order you want them heard. Since there will be only one
file, you know which will be first ;)

We actually do this with some of our queues, so I know it works.
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Re: [asterisk-users] Music On Hold

[Alexander Lopez]

Concatenate the files into one larger file, in the order you want them
to play

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Joel Hill
> Sent: Wednesday, September 26, 2007 7:01 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Music On Hold
> 
> Thanks for the suggestion, but I need it to play multiple messages.
> Always starting with the same one.
> 
> Cheers,
> 
> Joel.
> 
> On Wed, 2007-09-26 at 09:36 -0400, Forrest Beck wrote:
> > Make the file the only one in the /var/lib/asterisk/moh directory.
> >
> > Forrest Beck
> > [EMAIL PROTECTED]
> > www.shift8.biz
> >
> >
> >
> >
> >
> > On Sep 26, 2007, at 3:07 AM, Joel Hill wrote:
> >
> > > Hi All,
> > >
> > >
> > > I need to have the same file played from MoH every time someone
gets
> > > to
> > > MoH from a Dial. I want to play marketing messages from it and I
> > > want it
> > > to start from file 1 every time.
> > >
> > >
> > > Anyone know if/how this can be done?
> > >
> > >
> > > Cheers,
> > >
> > >
> > > Joel.
> > >
> > >
> > >
> > >
> > > ___
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Re: [asterisk-users] Music On Hold

[Wayne]

Hiya all,
Please excuse me if I'm a bit out of date with my Asterisk version here 
but... :)

I have noticed that the moh will start from where it left off from the 
previous caller, not from the beginning of the sound file. So going back 
to what Joal asked originally, having one file will mean that - yes 
things will be played in the correct order - but as each caller gets the 
moh - it would 'carry on' at the position it was last at when the 
/previous/ caller left moh - not from the start.


Cheery
Wayne.



David Gomillion wrote:
> > > Hi All,
> > >
> > > I need to have the same file played from MoH every time someone gets
> > > to
> > > MoH from a Dial. I want to play marketing messages from it and I
> > > want it
> > > to start from file 1 every time.
> > >
> > > Anyone know if/how this can be done?
>
> On Wed, 2007-09-26 at 09:36 -0400, Forrest Beck wrote:
> > Make the file the only one in the /var/lib/asterisk/moh directory.
> >
> > Forrest Beck
> > [EMAIL PROTECTED] 
> > www.shift8.biz 
> Thanks for the suggestion, but I need it to play multiple messages.
> Always starting with the same one.
>
> Cheers,
>
> Joel.
>
>
> Create a new MOH class with one large file consisting of every message 
> you want heard, in the order you want them heard. Since there will be 
> only one file, you know which will be first ;)
>
> We actually do this with some of our queues, so I know it works.
>
> 
>
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Re: [asterisk-users] Music On Hold

[Tilghman Lesher]

On Thursday 27 September 2007 17:00:33 Wayne wrote:
> I have noticed that the moh will start from where it left off from the
> previous caller, not from the beginning of the sound file. So going back
> to what Joal asked originally, having one file will mean that - yes
> things will be played in the correct order - but as each caller gets the
> moh - it would 'carry on' at the position it was last at when the
> /previous/ caller left moh - not from the start.

That's true if you use mpg123 for MOH... that's the old way.  The recommended
method now is to use native file format, which is saved per channel.  So every
channel gets the message started from the beginning.

-- 
Tilghman

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Re: [asterisk-users] Music On Hold

[Wayne]

Tilghman Lesher wrote:
> That's true if you use mpg123 for MOH... that's the old way.  The recommended
> method now is to use native file format, which is saved per channel.  So every
> channel gets the message started from the beginning.
>
>   
Aah - cheers for that :) I havnt updated in a while I must admit - must 
get round to having a looksee :)


Wayne.


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[asterisk-users] Music on Hold -- Error

[Ryan M. Colbert]

We use Asterisk 1.4.7 on CentOS with Bandwidth.com as our provider and Polycom 
330's for endpoints.  When one of our end points places a call on hold we get 
the following in CLI.  There is no music on hold provided for the caller.  The 
SIP.CONF entry for our connection to bandwithd.com specifies disallow=all and 
allow=ulaw.  Should there be a similar setting on the user.conf entries?

An interesting note is the IP noted in the CLI message below is neither 
Bandwidth.com nor the end point.

Thanks for any help!!

CLI Message:
[Nov 15 13:21:38] WARNING[11327]: channel.c:2964 set_format: Unable to find a 
codec translation path from ulaw to unknown
[Nov 15 13:21:38] WARNING[11327]: res_musiconhold.c:702 moh_alloc: Unable to 
set channel 'SIP/4.68.250.148-08d1e7e0' to format 'unknown'

Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue & McLain, P.A.
201 E. Pine Street, Suite 1500
Orlando, FL 32801
(407) 517-3105 - Direct Telephone
(407) 839-0120 - Main Office
(407) 841-9726 - Fax
http://www.rissman.com/

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Re: [asterisk-users] Music On Hold

[Godson Gera]

On Dec 18, 2007 3:58 AM, itgasterisk <[EMAIL PROTECTED]> wrote:

> Hello everyone,
>
> I am having a bit of problem getting MusicOnhold to play.
>
> I am running Asterisk 1.4 with MPG123 0.59 installed.
>
> And here's what i see in the debugging window of asterisk:
>
>-- Started music on hold, class 'default', on channel
> 'SIP/x123-082043d0'
>-- Stopped music on hold on SIP/x123-082043d0
>
> Any idea why it is not playing the file at all?
>
>
Hi Eric,

Try to install asterisk-addons which can play mp3 (using format_mp3.so)
files directly, instead of depending on mpg123. Once you install addons
don't forget to set mode=files in musiconhold.conf

-- 
Godson Gera,
http://godson.in
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Re: [asterisk-users] Music On Hold

[Tzafrir Cohen]

On Mon, Dec 17, 2007 at 05:28:12PM -0500, itgasterisk wrote:
> Hello everyone,
> 
> I am having a bit of problem getting MusicOnhold to play.
> 
> I am running Asterisk 1.4 with MPG123 0.59 installed.

Any specific reason you want to use mp3 format?

If you downsample this to a 8kHz 16 bits per sample mono wav file you'll
get a file which may be even smaller and will not take any special
transcoding to play by Asterisk on each time.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Music On Hold

[Patrick]

On Tue, 2007-12-18 at 13:28 +0530, Godson Gera wrote:
> 
> 
> On Dec 18, 2007 3:58 AM, itgasterisk <[EMAIL PROTECTED]>
> wrote:
> Hello everyone,
> 
> I am having a bit of problem getting MusicOnhold to play.
> 
> I am running Asterisk 1.4 with MPG123 0.59 installed.
> 
> And here's what i see in the debugging window of asterisk:
> 
>-- Started music on hold, class 'default', on channel 
> 'SIP/x123-082043d0'
>-- Stopped music on hold on SIP/x123-082043d0
> 
> Any idea why it is not playing the file at all?
> 
> 
> Hi Eric,
> 
> Try to install asterisk-addons which can play mp3 (using
> format_mp3.so) files directly, instead of depending on mpg123. Once
> you install addons don't forget to set mode=files in musiconhold.conf

Even better, don't use mp3 at all. Iirc variable bit rate mp3s could
cause Asterisk to blow up. Don't know if that has been fixed but do you
want to run that risk with a PBX? Just convert your music on hold files
to the native formats you use like ulaw, alaw, gsm, g729 etc. and
configure musiconhold.conf to use those files.

Regards,
Patrick


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[asterisk-users] Music on hold 1.2

[Ed Nuñez]

What is a good solution for playing music on hold on the 1.2 branch.  I do not 
want to use mpg123 because last time I used it in a production server it caused 
many problems.   The MPG123 process was taking about 60% of my Xeon CPU.

 

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[asterisk-users] Music on hold - 1.4.5

[Ade Vickers]

Hi,

Please bear with me if I'm asking stupid questions... I'm new to Asterisk,
newish to Linux, etc...

I've got MoH working nicely with my new Asterisk setup using the "files"
option; except that it always plays from the start of a (random) music file
when you first put someone on hold. Take them off hold & put them back, and
sometimes (not always!) it will start playing a new file from the
beginning If I park a call, from the point of pressing the "TRNF" button
the caller gets music; but, when the call parks, the music starts a new
file!

What I'd like to do is have the music streaming constantly, so the "on hold"
caller always gets music at the current position; even if that's in the
middle or near the end of a file.

The musiconhold.conf file mentions a couple of streaming options; but
(rightly) doesn't go into particular detail. So, what's my best strategy?

For info:
  - Asterisk is running on a P3 1GHz server (it's only a tiny experimental
PBX setup though)
  - v1.4.5, compiled by myself (thanks to voip-info.org & a couple of other
sites)
  - Server is Ubuntu "Fiesty Fawn", clean install (especially for Asterisk)
  - VoIP (SIP) only
  - All music files are in uLaw format, and the SIP phones are forced to use
uLaw encoding.

Cheers!
Ade.

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.476 / Virus Database: 269.9.12/878 - Release Date: 28/06/2007
17:57
 



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[asterisk-users] Music on hold problem

[yonoko molomo]

Hi,
I am using asterisk 1.4.
I have confgured the musiconhold.conf file.
However, when i make a call and then hold the call it does nothing.
in the CLI i do not see the starting/stopping musiconhold messages.

i am making calls from sip to h323 using asterisk assip/h323 gateway
(with gnugk and ooh323).
i get the following messages when putting the call on hold:

-- Executing [EMAIL PROTECTED]:1] Ringing("OOH323/1169fed2-70c5", "") in new 
stack
-- Executing [EMAIL PROTECTED]:2] Dial("OOH323/1169fed2-70c5",
"SIP/ht04|6|r") in new stack
-- Called ht04
-- SIP/ht04-081fe7f0 is ringing
-- SIP/ht04-081fe7f0 answered OOH323/1169fed2-70c5
<...on hold...>
[Jul 17 11:19:22] WARNING[23645]: src/chan_h323.c:1044
ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_1
[Jul 17 11:19:25] NOTICE[23645]: rtp.c:783 process_rfc3389: Comfort
noise support incomplete in Asterisk (RFC 3389). Please turn off on
client if possible. Client IP: 10.4.0.116
[Jul 17 11:19:25] WARNING[23645]: src/chan_h323.c:1044
ooh323_indicate: Don't know how to indicate condition 16 on ooh323c_1

I have no idea why the musiconhold is not triggered,  what those
messages mean (dont know how to indicate condition) and if they are
related to the music on hold problem


someone has any idea?

in the CLI i type
CLI> moh show files
and i see one file i put in the directory.

i tried configuring the musiconhold.conf file as quitemp3, files and
also custom (installing mpg123) but none of them starts musiconhold.

do i need to activate musiconhold somewhere else?

any help is welcome
thanks

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Re: [asterisk-users] music on hold

[Lee, John (Sydney)]

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 

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Re: [asterisk-users] music on hold

[Tzafrir Cohen]

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

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Re: [asterisk-users] music on hold

[邱磊]

^_^
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 ^^;


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Re: [asterisk-users] music on hold

[邱磊]

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 
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Re: [asterisk-users] music on hold

[Jeff LaCoursiere]

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

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Re: [asterisk-users] music on hold

[fateme fatah]

See:
http://astrecipes.net/index.php?q=AstRecipes/Music-on-hold%20without%20MPG123


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Re: [asterisk-users] Music On Hold

[Danny Nicholas]

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

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Re: [asterisk-users] Music On Hold

[Ex Vito]

On Fri, Jan 30, 2009 at 3:23 PM, Danny Nicholas  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

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Re: [asterisk-users] Music On Hold

[Idris AVCI]

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  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

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Re: [asterisk-users] Music On Hold

[Ex Vito]

On Mon, Feb 2, 2009 at 8:39 AM, Idris AVCI  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

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[asterisk-users] music on hold realtime

[Nhadie]

Hi,

Is it possible to use realtime for Music On Hold?
Is it also possible to store the music/audio files on the database, same
way a voicemail can be stored on the database?

Thank You

Regards,
Nhadie


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Re: [asterisk-users] Music on hold

[Vazquez David]

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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Re: [asterisk-users] Music on Hold

[Brent Davidson]

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.

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Re: [asterisk-users] Music on Hold

[Julien Claassen]

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

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Re: [asterisk-users] Music on Hold

[Danny Nicholas]

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.

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Re: [asterisk-users] Music on Hold

[Dan Saul]

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  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.
>
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Re: [asterisk-users] Music on Hold

[Miguel Molina]

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


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Re: [asterisk-users] Music on Hold

[Danny Nicholas]

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  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.


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Re: [asterisk-users] Music on Hold

[Dan Saul]

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  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  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