Hello,
I have been trying to completely disable music on hold on my asterisk
system. When a call is put on hold I do not want any music on hold, but I
would like the remote user to get informed of this event (depending on the
technology e.g. with a SIP reinvite and an SDP indicating the call is on
hold).
I have searched and tried out various approaches, but when putting the
call on hold by a SIP user, I always get an indication that asterisk tries
to play music on hold. The remote side does not get a re-invite.
What I have tried so far:
- no musiconhold.conf in the hope that lack of the configuration file
disables moh
- a musiconhold.conf where everything is commented out
- modules.conf with 'unload => res_musiconhold.so'
When I start asterisk, it indicates that it disables music on hold:
[Jan 28 10:15:02] WARNING[31052]: res_musiconhold.c:1784 load_module: No
music on hold classes configured, disabling music on hold.
== Registered application 'MusicOnHold'
== Registered application 'WaitMusicOnHold'
== Registered application 'SetMusicOnHold'
== Registered application 'StartMusicOnHold'
== Registered application 'StopMusicOnHold'
res_musiconhold.so => (Music On Hold Resource)
However, when I set up a sip call between two sip phones and one end puts
the call on hold, then I always get the following message and the remote
side is not informed that the call is on hold:
-- Executing [s at macro-stddial:2] Dial("SIP/2222-00000000",
"SIP/4444")
in new stack
== Using SIP RTP CoS mark 5
-- Called 4444
-- SIP/4444-00000001 is ringing
-- SIP/4444-00000001 answered SIP/2222-00000000
-- Native bridging SIP/2222-00000000 and SIP/4444-00000001
later when the call is put on hold:
-- Music class default requested but no musiconhold loaded.
Can anybody give me any pointers or tell me how to disable moh completely
and send re-invites for call hold?
thanks for any help
Urs
My easiest configuration with Asterisk 1.6.2.7:
modules.conf
----------------------------------------------------------
[modules]
autoload=yes
; res_phoneprov requires func_strings.so to be loaded:
preload => func_strings.so
noload => pbx_gtkconsole.so
noload => res_musiconhold.so
extensions.conf:
-----------------------------------------------------------
[general]
[default]
;SIP extensions
exten => _XXXX,1,Macro(stddial,SIP/${EXTEN})
[macro-stddial]
; ${ARG1} - What to dial
exten => s,1,Answer()
exten => s,n,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => _s-.,1,NoOp
sip.conf:
--------------------------------------------------------
[general]
language=en ; configured default language
dtmfmode=rfc2833 ; default dtmfmode for sending DTMF
(Dual-tone multi-frequency)
directrtpsetup=no ; Disable the new experimental direct
RTP setup
allowtransfer=yes ; enable all transfers for peers and
users
match_auth_username=yes ; use 'authentication username'
instead of 'username for authentication' (if available)
session-timers=originate ; Request and run session-timers
always
session-expires=3600 ; maximum session refresh interval
session-minse=600 ; minimum session refresh interval
session-refresher=uas ; session refresher is
user-agent-server
;allowguest=no ; Allow or reject guest calls (default
is yes)
notifyhold = yes ; Notify subscriptions on HOLD state
(default: no)
udpbindaddr=0.0.0.0:5060 ; Servers IP address (all) to bind UDP
listen socket to
srvlookup=yes ; enable DNS SRV lookups on outbound
calls
[allusers](!)
context=default
type=friend ; All options are possible
qualify=2000 ; no, 2000=2 sec to wait for reply
before remote party is considered unreachable
;qualifyfreq=60 ; Qualification: How often to check in
seconds
canreinvite=yes ; certain devices do not like change
of RTP source/destination during call
[4444](allusers)
host=dynamic ; the device needs to register
secret = 1234
[3333](allusers)
host=dynamic ; the device needs to register
secret = 1234
[2222](allusers)
host=dynamic ; the device needs to register
secret = 1234
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