[asterisk-users] Packet2packet bridging while in sip.conf canreinvite=no

[jonas kellens]

I have put canreinvite=no for all my internal SIP-clients in sip.conf
because I want Asterisk to be in the middle of the RTP-stream so he can
provide MusiconHold and so...

Now, what the Asterisk CLI tells me when I make a call from my one
internal SIP-phone to another internal SIP-phone is :

Verbosity is at least 25
  == Spawn extension (intern, 51, 1) exited non-zero on
'SIP/BT201-088f93e0'
    -- Executing [52 at intern:1] Dial("SIP/GXP1200-088f93e0", "SIP/BT201|
30") in new stack
    -- Called BT201
    -- SIP/BT201-088faa00 is ringing
    -- SIP/BT201-088faa00 answered SIP/GXP1200-088f93e0
    -- Packet2Packet bridging SIP/GXP1200-088f93e0 and
SIP/BT201-088faa00
  == Spawn extension (intern, 52, 1) exited non-zero on
'SIP/GXP1200-088f93e0'

Why is there this native bridging ? Does this mean that Asterisk is no
longer in the middle of it ?

Also : there is no audio at all ! Just when I put down the phone there's
the DTMF-signal that the line is cancelled...

Everything worked well before I edited musiconhold.conf and
features.conf (to create a park extension).


My sip.conf :

[root at asterisk asterisk]# cat sip.conf
[general]
context=default
port=5060
bindaddr=192.168.4.248
srvlookup=yes
disallow=all
allow=alaw
allow=gsm
allow=ulaw
language=be

[BT201]
type=friend
context=intern
host=dynamic
username=BT201
secret=testpaswoord
canreinvite=no
callerid=Jonas Kellens <52>
qualify=yes

[GXP1200]
type=friend
context=intern
host=dynamic
username=GXP1200
secret=testpaswoord
canreinvite=no
callerid=callerid? <51>
qualify=yes


[GXP2020]
type=friend
context=intern
host=dynamic
username=GXP2020
secret=testpaswoord
canreinvite=no
callerid=Kristof Teirlinck <50>
qualify=yes

Musiconhold.conf :

[default]
mode=files
directory=/var/lib/asterisk/moh
random=yes

Features.conf :

; Sample Call Features (parking, transfer, etc) configuration
;

[general]
parkext => 90			; What extension to dial to park
parkpos => 91-95		; What extensions to park calls on. These needs to be
				; numeric, as Asterisk starts from the start position
				; and increments with one for the next parked call.
context => parkedcalls		; Which context parked calls are in
parkingtime => 60		; Number of seconds a call can be parked for 
				; (default is 45 seconds)
;courtesytone = beep		; Sound file to play to the parked caller 
				; when someone dials a parked call
				; or the Touch Monitor is activated/deactivated.
;parkedplay = caller		; Who to play the courtesy tone to when picking up
a parked call
				; one of: parked, caller, both  (default is caller)
;parkedcalltransfers = caller   ; Enables or disables DTMF based
transfers when picking up a parked call.
                                ; one of: callee, caller, both, no
(default is both)
;parkedcallreparking = caller   ; Enables or disables DTMF based
one-touch parking when picking up a parked call.
                                ; one of: callee, caller, both, no
(default is no)
;parkedcallhangup = caller      ; Enables or disables DTMF based hangups
when picking up a parked call.
                                ; one of: callee, caller, both, no
(default is no)
;parkedcallrecording = caller   ; Enables or disables DTMF based
one-touch recording when picking up a parked call.
                                ; one of: callee, caller, both, no
(default is no)
;adsipark = yes			; if you want ADSI parking announcements
;findslot => next		; Continue to the 'next' free parking space. 
				; Defaults to 'first' available
parkedmusicclass=default	; This is the MOH class to use for the parked
channel
				; as long as the class is not set on the channel directly
				; using Set(CHANNEL(musicclass)=whatever) in the dialplan

;transferdigittimeout => 3	; Number of seconds to wait between digits
when transferring a call
				; (default is 3 seconds)
;xfersound = beep		; to indicate an attended transfer is complete
;xferfailsound = beeperr	; to indicate a failed transfer
pickupexten = *8		; Configure the pickup extension. (default is *8)
;featuredigittimeout = 1000 ; Max time (ms) between digits for 
                            ; feature activation  (default is 1000 ms)
;atxfernoanswertimeout = 15 ; Timeout for answer on attended transfer
default is 15 seconds.


Do you need extra info ??
What setting can I have set in musiconhold.conf or features.conf to
affect the audiostream between my clients ???

Before I could call all my clients, I had musiconhold when putting 'on
hold' and I was just figuring out how parked calls worked...

Thanks for the help !

Jonas Kellens.
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