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</head><BODY ><div><SPAN
style="font-family:'Arial';font-size:10pt;">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.</SPAN></div>
<div> </div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;">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.</SPAN></div>
<div> </div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;">Here is what
the console said:</SPAN></div>
<div> </div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;">[Jun 12
08:27:16] WARNING[23414]: channel.c:2947 set_format: Unable to find a codec
translation path from ulaw to g729</SPAN></div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;">[Jun 12
08:27:16] WARNING[23414]: channel.c:2947 set_format: Unable to find a codec
translation path from ulaw to g729</SPAN></div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;"> -- Stopped
music on hold on SIP/5000-b6c013c8</SPAN></div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;">[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)</SPAN></div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;">[Jun 12
08:27:16] WARNING[23504]: channel.c:4215 ast_channel_bridge: Can't make
SIP/5000-b6c013c8 and SIP/5003-08263798 compatible</SPAN></div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;">[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</SPAN></div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;"> -- adaptive
jitterbuffer destroyed on channel SIP/5003-08263798</SPAN></div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;"> == Spawn
extension (macro-stdexten, s, 6) exited non-zero on 'SIP/5000-b6c013c8'
in macro 'stdexten'</SPAN></div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;"> == Spawn
extension (macro-stdexten, s, 6) exited non-zero on
'SIP/5000-b6c013c8'</SPAN></div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;"> -- adaptive
jitterbuffer destroyed on channel SIP/5000-b6c013c8</SPAN></div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;"> -- adaptive
jitterbuffer destroyed on channel SIP/5004-0828b298</SPAN></div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;"> == Spawn
extension (macro-page, s, 6) exited non-zero on 'SIP/5003-b6c05830' in
macro 'stdexten'</SPAN></div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;"> == Spawn
extension (macro-page, s, 6) exited non-zero on
'SIP/5003-b6c05830'</SPAN></div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;"> -- adaptive
jitterbuffer destroyed on channel SIP/5003-b6c05830</SPAN></div>
<div> </div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;">Do I really
need a license to bridge G729 RTP traffic on Asterisk 1.4? Why is it trying to
go to ulaw?</SPAN></div>
<div> </div>
<div><SPAN
style="font-family:'Arial';font-size:10pt;">stdexten macro
has the following dial command:</SPAN></div>
<div> </div>
<div><SPAN style="font-family:'Courier
New';font-size:9pt;">exten =>
s,n(dial),Dial(${ARG2},20)
; Ring the interface, 20 seconds maximum</SPAN></div>
<div><SPAN style="font-family:'Courier
New';font-size:9pt;"><br />where ARG2 is the device to
ring.</SPAN></div>
<div> </div>
<div> </div>
</body></html>
On 6/12/07, Tony Plack <Tony@plack.net> wrote:> > 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. >I have seen phones that only allow ONE g.729 stream to be operating at a time. You may want to check the documentation and see if that's what the licensing on the GXP-2000 allows. Hope that helps nudge you in the right direction, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/da689e3e/attachment.htm
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@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.-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/e51be958/attachment.htm