asterisk users - Mar 2007 - How to enter bridge_native_loop???

Hi,

I am using asterisk-1.4.0.

I am inquisitive of what Packet2Packet bridge (bridge_p2p_loop) does and
what Native bridge (bridge_native_loop) does.

I have configured my dial plans and options such that I can enter
bridge_p2p_loop. However, I am unable to enter bridge_native_loop for some
reason.

I have the following extensions:

exten => 7126,1,Dial(SIP/lin_santosh)
exten => 7126,s+1,Hangup

exten => 7140,1,Dial(SIP/win_test)
exten => 7140,s+1,Hangup

My sip.conf is as:

[lin_santosh]
type=friend
regexten=7126
callerid="LIN Santosh" <7126>
host=dynamic
nat=yes
canreinvite=no
allow=all

[win_test]
type=friend
regexten=7140
callerid="WIN Test" <7140>
host=dynamic
nat=yes
canreinvite=no
allow=all


My features.conf has:
[featuremap]
;blindxfer => #         ; Blind transfer  (default is #)
;disconnect => *0               ; Disconnect  (default is *)
;automon => *1                  ; One Touch Record a.k.a. Touch Monitor
;atxfer => *                    ; Attended transfer
;parkcall => #72                ; Park call (one step parking)

Should this not be enough for bridging to happen via bridge_native_loop?

Am I missing something?

Thanks in anticipation.

Regards,
Santosh.
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9 mar 2007 kl. 08.52 skrev Santosh Raghuram:
> Hi,
>
> I am using asterisk-1.4.0.
>
> I am inquisitive of what Packet2Packet bridge (bridge_p2p_loop)  
> does and what Native bridge (bridge_native_loop) does.
>
> I have configured my dial plans and options such that I can enter  
> bridge_p2p_loop. However, I am unable to enter bridge_native_loop  
> for some reason.
>
> I have the following extensions:
>
> exten => 7126,1,Dial(SIP/lin_santosh)
> exten => 7126,s+1,Hangup
>
> exten => 7140,1,Dial(SIP/win_test)
> exten => 7140,s+1,Hangup
>
> My sip.conf is as:
>
> [lin_santosh]
> type=friend
> regexten=7126
> callerid="LIN Santosh" <7126>
> host=dynamic
> nat=yes
> canreinvite=no
> allow=all
>
You have set canreinvite to no, thus disabling native briding.

/O



---
* Olle E. Johansson - oej@edvina.net
* Asterisk Training http://edvina.net/training/
* Asterisk SIP Masterclass - Stockholm, Sweden, May 7-11

Hi,

With canreinvite=yes, all the media/rtp traffic for the call typically flows
directly between the two peers. So how is the code in bridge_native_loop
called and when? Is it called and used for any further sip signalling and
not rtp?


Thanks for your prompt reply.

Regards,
Santosh.
> Hi,
>
> I am using asterisk-1.4.0.
>
> I am inquisitive of what Packet2Packet bridge (bridge_p2p_loop)
> does and what Native bridge (bridge_native_loop) does.
>
> I have configured my dial plans and options such that I can enter
> bridge_p2p_loop. However, I am unable to enter bridge_native_loop
> for some reason.
>
> I have the following extensions:
>
> exten => 7126,1,Dial(SIP/lin_santosh)
> exten => 7126,s+1,Hangup
>
> exten => 7140,1,Dial(SIP/win_test)
> exten => 7140,s+1,Hangup
>
> My sip.conf is as:
>
> [lin_santosh]
> type=friend
> regexten=7126
> callerid="LIN Santosh" <7126>
> host=dynamic
> nat=yes
> canreinvite=no
> allow=all
>
You have set canreinvite to no, thus disabling native briding.

/O
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> Remember that each channel/media driver has different ways to handle
> this.
> Zapata has native bridging too, but different rules apply depending on
> the
> technology used.
>
> /Olle
Thanks Olle for your help, the details were exhaustive.

Regards,
Santosh.
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