asterisk users - Apr 2009 - Packet2packet bridging while in sip.conf canreinvite=no

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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jonas kellens wrote:
> 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 ?
It is important to note that Packet2Packet bridging is not the same as native 
bridging. With native bridging, the audio flows outside of Asterisk between the 
endpoints. With P2P bridging, the audio comes into the RTP layer of Asterisk but
does not pass through the Asterisk core. This allows for Asterisk to intercept 
DTMF or play warning files to the bridged parties.
> 
> Also : there is no audio at all ! Just when I put down the phone
there's
> the DTMF-signal that the line is cancelled...
SIP debug would probably help.
> 
> Everything worked well before I edited musiconhold.conf and 
> features.conf (to create a park extension).
Looking at your musiconhold.conf file, it looks very much like the sample 
musiconhold.conf file. I doubt that your changes there would have affected 
anything. If you say that the problems started when you edited features.conf, 
then I would suggest that you start undoing the changes you made one-by-one to 
see if you can find what change it was that caused the problem to occur.

[sample configs snipped]
> 
> Do you need extra info ??
> What setting can I have set in musiconhold.conf or features.conf to 
> affect the audiostream between my clients ???
There is nothing you can set in musiconhold.conf to control the media stream. 
With SIP, the signalling still goes through Asterisk even if the media does not.
  Even if Asterisk is not in the media path, the endpoints can still signal to 
Asterisk to play MOH to the other side. Asterisk can accomplish this through 
reinvites.

Also, there is nothing you can set in features.conf to control the media stream.
Settings pertaining to the media stream are channel-driver-specific and are thus
configured in each particular channel driver's configuration file. As you
have
already discovered, the setting which forces media onto Asterisk during a SIP 
call is the canreinvite setting.

Mark Michelson
> 
> 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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