-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all!
I am trying to redirect to a local extension the incoming calls that I receive to an account which I have in iptel.org, but when receiving I'm obtaining this error: alderamin*CLI> -- Executing [...@from-internal:1] Dial("SIP/danib-089f8820", "SIP/300|30|tTrm") in new stack [Feb 19 19:22:50] WARNING[19254]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) [Feb 19 19:23:00] WARNING[19254]: pbx.c:2529 __ast_pbx_run: Timeout, but no rule 't' in context 'from-internal' It is probable that this can be due to a problem of interaction between contexts? I copy the content of extensions.conf and sip.conf to see if it can help to find the problem: - ------------------------------------------------------------------------ extensions.conf: ; DGB - 20091114 [general] autofallthrough=no [macro-dial] exten => s,1,Dial(${ARG1},15) exten => s,n,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(${macro_ext...@voicemail,u) exten => s-NOANSWER,n,Hangup exten => s-BUSY,1,Voicemail(${macro_ext...@voicemail,b) exten => s-BUSY,n,Hangup exten => s-CHANUNAVAIL,1,Playback(pbx-invalid) [from-internal] ; Llamadas a extensiones SIP exten => _2xx,1,Macro(dial,SIP/${EXTEN}) exten => _2xx,n,Hangup exten => 300,1,Dial(SIP/300,30,tTrm) ; Extension analogica exten => 402,1,Macro(dial,DAHDI/2) exten => 402,n,Hangup ; Directorio de extensiones exten => *400,1,Directory(voicemail,from-internal) ; Musica en espera exten => *300,1,Answer exten => *300,n,SetMusicOnHold(default) exten => *300,n,WaitMusicOnHold(2000) exten => *300,n,Hangup ; Prueba de Eco exten => *200,1,Answer exten => *200,n,Playback(demo-echotest) exten => *200,n,Echo exten => *200,n,Playback(demo-echodone) exten => *200,n,Hangup ; Acceso a voicemail exten => *100,1,Answer exten => *100,n,Wait(1) exten => *100,n,VoiceMailMain(${CALLERID(num)}...@voicemail) exten => *100,n,Hangup ; Llamadas salientes exten => _9.,1,Dial(DAHDI/1/${EXTEN:1}) exten => _9.,n,Hangup ; Call a number at iptel.org exten => _0.,1,Dial(SIP/iptel/${EXTEN:1},20,r)) exten => _0.,n,Hangup [from-pstn] ; incoming calls from FXO port are directed to this context exten => s,1,Answer() exten => s,n,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=15) exten => s,n,Background(contestador1) exten => i,1,Goto(from-pstn,s,1) exten => t,1,Playback(locomunicoconelinterno1) exten => t,n,Dial(SIP/200,25) exten => t,n,VoiceMail(2...@voicemail,20) exten => t,n,Hangup() include => from-internal - ------------------------------------------------------------------------ sip.conf: [general] [...] ; register with iptel.org register => danib:mlrzv...@iptel.org/300 [...] ; Outgoing to iptel.org [iptel] type=friend username=danib secret=myspasswd host=iptel.org canreinvite=no qualify=300 insecure=port,invite ; required for incoming ekiga.net calls context = from-internal - ------------------------------------------------------------------------ Thanks in advance for your replies. Regards, Daniel -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkt/LkUACgkQZpa/GxTmHTeglwCgh8E59wZ+9yBXEWhwC+RdnZgP 16MAnRh4NDaN9QOGHjIRbvWUQtiA2v23 =6iU8 -----END PGP SIGNATURE----- -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users