[asterisk-users] Bridge bug in 1.4?

[rjcarvalho at reit.up.pt]

Hi Tony,

Since G.729 codec requires a license unless using pass-thru, normal  
calls probably pass without transcoding in your server to the other  
end, but when transferring calls, Asterisk needs to transcode the RTP  
flows, and that needs to be licensed!

You can however use other codec on every phone you got, and also in  
the list of codec capabilities exchanged with your provider too, and  
like this, on every call,G.729 never gets selected, and transfers will  
work just fine!

   Regards,
Ricardo.

   Quoting Tony Plack <Tony at plack.net>:

>    I have GXP-2000 phones running against Asterisk 1.4.  All phones are
> running G729 and this is witnessed by the fact that the phone shows
> the G729 codec.   I dial the first phone, place it on hold, dial the
> second phone, press CONF and the other line.  The first connection
> goes away and the second remains connected.   Here is what the
> console said:   [Jun 12 08:27:16] WARNING[23414]: channel.c:2947
> set_format: Unable to find a codec translation path from ulaw to g729
> [Jun 12 08:27:16] WARNING[23414]: channel.c:2947 set_format: Unable to
> find a codec translation path from ulaw to g729     -- Stopped music
> on hold on SIP/5000-b6c013c8 [Jun 12 08:27:16] WARNING[23504]:
> channel.c:3288 ast_channel_make_compatible: No path to translate from
> SIP/5000-b6c013c8(256) to SIP/5003-08263798(4) [Jun 12 08:27:16]
> WARNING[23504]: channel.c:4215 ast_channel_bridge: Can't make
> SIP/5000-b6c013c8 and SIP/5003-08263798 compatible [Jun 12 08:27:16]
> WARNING[23504]: res_features.c:1458 ast_bridge_call: Bridge failed on
> channels SIP/5000-b6c013c8 and SIP/5003-08263798     -- adaptive
> jitterbuffer destroyed on channel SIP/5003-08263798   == Spawn
> extension (macro-stdexten, s, 6) exited non-zero on
> 'SIP/5000-b6c013c8' in macro 'stdexten'   == Spawn extension
> (macro-stdexten, s, 6) exited non-zero on 'SIP/5000-b6c013c8'     --
> adaptive jitterbuffer destroyed on channel SIP/5000-b6c013c8     --
> adaptive jitterbuffer destroyed on channel SIP/5004-0828b298   ==
> Spawn extension (macro-page, s, 6) exited non-zero on
> 'SIP/5003-b6c05830' in macro 'stdexten'   == Spawn extension
> (macro-page, s, 6) exited non-zero on 'SIP/5003-b6c05830'     --
> adaptive jitterbuffer destroyed on channel SIP/5003-b6c05830   Do I
> really need a license to bridge G729 RTP traffic on Asterisk 1.4?
>  Why is it trying to go to ulaw?   stdexten macro has the following
> dial command:   exten => s,n(dial),Dial(${ARG2},20)            ; Ring
> the interface, 20 seconds maximum
> where ARG2 is the device to ring.




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