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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>
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<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>
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<div><SPAN style="font-family:'Arial';font-size:10pt;">Here is what the console said:</SPAN></div>
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<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>
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<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>
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<div><SPAN style="font-family:'Arial';font-size:10pt;">stdexten macro has the following dial command:</SPAN></div>
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<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>
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