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Home > Archive > Voice Over IP in UK > January 2007 > music on hold by preesing button "R"
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music on hold by preesing button "R"
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| gvido.sprogis@gmail.com 2007-01-10, 7:11 am |
| HI! It's me again.
Now my problem is: I make a call from Farfisa through asterisk to SPA
3000 FXO port. when I press 2 times button "R" on the attached phone,
asterisk asterisk is sending DTMF signal to Farfisa.
I make another call from Farfisa to .... FXO port. when I press butto
"R" tha asterisk
O, and very interesting thing:
when I pick up the phone after first ring->
-- Executing Dial("SIP/300-8512", "SIP/303|15") in new stack
-- Called 303
-- SIP/303-131a is ringing
-- SIP/303-131a answered SIP/300-8512
-- Attempting native bridge of SIP/300-8512 and SIP/303-131a
-- Started music on hold, class 'default', on SIP/300-8512
-- Stopped music on hold on SIP/300-8512
-- Started music on hold, class 'default', on SIP/300-8512
-- Stopped music on hold on SIP/300-8512
== Spawn extension (internal, 193, 2) exited non-zero on
'SIP/300-8512'
when I pick up the phone after second ring ->
-- Executing Dial("SIP/300-8512", "SIP/303|15") in new stack
-- Called 303
-- SIP/303-131a is ringing
-- SIP/303-131a answered SIP/300-8512
-- Attempting native bridge of SIP/300-8512 and SIP/303-131a
-- Started music on hold, class 'default', on SIP/300-8512
-- Stopped music on hold on SIP/300-8512
-- Started music on hold, class 'default', on SIP/300-8512
-- Stopped music on hold on SIP/300-8512
== Spawn extension (internal, 193, 2) exited non-zero on
'SIP/300-8512'
my features.conf
;
; Sample Parking configuration
;
[general]
parkext => 700
parkpos => 701-720
context => parkedcalls
parkingtime => 3600
;transferdigittimeout => 3
xfersound = beep
xferfailsound = beeperr
adsipark = yes
;findslot => next
;automon => *1
[featuremap]
disconnect => *0
automon => *1
atxfer => *2
blindxfer => #2
pickupexten = *8
[applicationmap]
;testfeature => #9,callee,Playback,tt-monkeys
my sip.conf for peer 193, 300 and 303
[193] ; Gvido linux on pc
type=friend
secret=1234
qualify=yes
nat=no
host=dynamic
context=internal
Callgroup=1
pickupgrou=1
dtmfmode=inband
disallow=all
allow=alaw
[300]; Farfisa
type=friend
secret=farfisa1234
qualify=yes
nat=no
host=dynamic
context=internal
Callgroup=1
pickupgroup=1
dtmfmode=inband
;disallow=all
;allow=ulaw
[303]; Sipura
type=friend
secret=sipura303
qualify=yes
nat=no
host=dynamic
context=internal
Callgroup=1
pickupgroup=1
dtmfmode=inband
;disalow=all
;allow=alaw
SPA 3000 Line 1
Audio Configuration
Preferred Codec:G726-32
Use Pref Codec Only:No
G729a Enable:Yes
G723 Enable:Yes
G726-16 Enable: Yes
G726-24 Enable: Yes
G726-32 Enable: Yes
G726-40 Enable: Yes
DTMF Process INFO:yes
DTMF Process AVT:NO
DTMF Tx Method: Auto
Hook Flash Tx Method: Info
SPA PSTN line
Audio Configuration
Preferred Codec:G726-32
Use Pref Codec Only:No
G729a Enable:Yes
G723 Enable:Yes
G726-16 Enable:Yes
G726-24 Enable: Yes
G726-32 Enable: Yes
G726-40 Enable: Yes
DTMF Process INFO:Yes
DTMF Process AVT:Yes DTMF Tx Method: Inband
PSTN line tab dial plan: (SO<:193@192.168.1.5> )
Line 1 tab enable IP dialing yes
could You please help me, again? Thank You =)
| |
| ale.cx 2007-01-10, 7:11 pm |
|
gvido.spro...@gmail.com wrote:
> HI! It's me again.
> Now my problem is: I make a call from Farfisa through asterisk to SPA
> 3000 FXO port. when I press 2 times button "R" on the attached phone,
> asterisk asterisk is sending DTMF signal to Farfisa.
What are you attempting to acheive? Put someone on hold and make
another call?
> I make another call from Farfisa to .... FXO port. when I press butto
> "R" tha asterisk
>
> O, and very interesting thing:
<snip>
I think something is missing in your post!
> [303]; Sipura
<snip>
> dtmfmode=inband
> SPA 3000 Line 1
<snip>
> DTMF Process INFO:yes
> DTMF Process AVT:NO
> DTMF Tx Method: Auto
> Hook Flash Tx Method: Info
They don't match! Make [303] use dtmfmode=info [like the SPA3000] or
vice versa, and you should be all set.
> could You please help me, again? Thank You =)
What is the problem you're having?
alexd
| |
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| ale.cx wrote:
> gvido.spro...@gmail.com wrote:
>
> What are you attempting to acheive? Put someone on hold and make
> another call?
>
> <snip>
>
> I think something is missing in your post!
>
> <snip>
>
> <snip>
>
> They don't match! Make [303] use dtmfmode=info [like the SPA3000] or
> vice versa, and you should be all set.
>
>
>
> What is the problem you're having?
>
> alexd
>
Sending a hook flash via SIP INFO probably wont work. I'd change the
hook flash Tx method to AVT.
cheers,
Paul.
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