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 <[EMAIL PROTECTED]>:
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.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
