Great, compiled and running<br><br>Only a few bugs:<br><br><br>A call to Musiconhold is OK. But imposible to hang up the channel.<br><br>A call to a sip endpoint fails about codec....<br><br>dial 1001<br> -- Executing [
1001@from-internal:1] NoOp("Console/default", "from-internal") in new stack<br> -- Executing [1001@from-internal:2] Set("Console/default", "CALLERID(all)=100147") in new stack<br>
-- Executing [1001@from-internal:3] Dial("Console/default", "SIP/adamvozip/1001|30|T") in new stack<br>[Jan 3 00:18:09] WARNING[91464]: channel.c:610 ast_best_codec: Don't know any of 0x8000 formats
<br>[Jan 3 00:18:09] WARNING[91464]: channel.c:610 ast_best_codec: Don't know any of 0x0 formats<br>Audio is at <a href="http://192.168.1.103">192.168.1.103</a> port 17678<br>Adding codec 0x400 (ilbc) to SDP<br>Adding codec 0x2 (gsm) to SDP
<br>Adding codec 0x8 (alaw) to SDP<br>Adding codec 0x4 (ulaw) to SDP<br>Adding codec 0x40 (slin) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br>Reliably Transmitting (NAT) to <a href="http://212.36.71.100:5060">
212.36.71.100:5060</a>:<br>INVITE <a href="mailto:sip:1001|30|T@sip.adamvozip.es">sip:1001|30|T@sip.adamvozip.es</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://192.168.1.103:5060">192.168.1.103:5060</a>;branch=z9hG4bK61e2ac31;rport
<br>Max-Forwards: 70<br>From: "100147" <<a href="mailto:sip:100147@sip.adamvozip.es">sip:100147@sip.adamvozip.es</a>>;tag=as442d100c<br>To: <<a href="mailto:sip:1001|30|T@sip.adamvozip.es">sip:1001|30|T@sip.adamvozip.es
</a>><br>Contact: <<a href="mailto:sip:100147@192.168.1.103">sip:100147@192.168.1.103</a>><br>Call-ID: <a href="mailto:4c632292789cbb804fabaeb07015b694@sip.adamvozip.es">4c632292789cbb804fabaeb07015b694@sip.adamvozip.es
</a><br>CSeq: 102 INVITE<br>User-Agent: AppleTV PBX<br>Date: Wed, 02 Jan 2008 23:18:09 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Content-Type: application/sdp<br>Content-Length: 391
<br><br>v=0<br>o=root 1727718394 1727718394 IN IP4 <a href="http://192.168.1.103">192.168.1.103</a><br>s=Asterisk PBX SVN-trunk-r96025<br>c=IN IP4 <a href="http://192.168.1.103">192.168.1.103</a><br>t=0 0<br>m=audio 17678 RTP/AVP 97 3 8 0 10 101
<br>a=rtpmap:97 iLBC/8000<br>a=fmtp:97 mode=30<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:10 L16/8000<br><br><br>Any suggesion?<br><br>*CLI> core show channel Console/default <br>
Console/default is not a known channel<br><br><br><br>*CLI> soft hangup Console/default <br>Console/default is not a known channel<br><br><br><br><br><div class="gmail_quote">On Jan 2, 2008 9:39 PM, Kevin P. Fleming <
<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div>
<div class="Wj3C7c">Adrià Vidal wrote:<br>> have $ svn co <a href="http://svn.digium.com/svn/asterisk/team/russell/chan_console" target="_blank">http://svn.digium.com/svn/asterisk/team/russell/chan_console</a><br>> dead?
<br>><br>> trying to test it into my macbook too...<br><br></div></div>It's been merged into SVN trunk already, you can just test the trunk<br>instead.<br><font color="#888888"><br>--<br>Kevin P. Fleming<br>Director of Software Technologies
<br>Digium, Inc. - "The Genuine Asterisk Experience" (TM)<br></font><div><div></div><div class="Wj3C7c"><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--" target="_blank">
http://www.api-digital.com--</a><br><br>asterisk-dev mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev
</a><br></div></div></blockquote></div><br><br clear="all"><br>-- <br>--<br>Adrià Vidal<br><a href="mailto:adriavidal@gmail.com">adriavidal@gmail.com</a>