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

[Mark Michelson]

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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