Pessoal... to tentando a um tempão a transferência de chamadas e ainda não consegui. Um pergunta: Tem como eu puxar uma ligação de um ramal?.<br>O q tenho é o seguinte:<br><font style="font-family: arial,sans-serif;" size="2">uso o trixbox trixbox CE current release is <a href="http://2.6.0.7">2.6.0.7</a><br>
</font><pre style="margin: 2px 0px 0px; font-family: arial,sans-serif;"><font size="2">Asterisk 1.4.18-3</font></pre><br><br>Edit: features_featuremap_additional.conf<br>transferdigittimeout => 10<br>featuredigittimeout = 3000<br>
<br>blindxfer=##<br>atxfer=*2<br>automon=*1<br>disconnect=**<br><br>ligo do 774 para o 711<br><br>no 711 eu aperto '#' e nao '##' e da o som de transferencia<br><br>disco 773 e da que o numero discado nao está em serviço no ramal 774 e no 711 da tom de ocupado<br>
<br>segue o log:<br><br><br> -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck<br> recordingcheck|20080506-082331|1210073011.26: Inbound recording not enabled<br> -- AGI Script recordingcheck completed, returning 0<br>
-- Executing [s@macro-record-enable:5] NoOp("SIP/774-093cfaf0", "No recording needed") in new stack<br> -- Executing [s@macro-exten-vm:9] Macro("SIP/774-093cfaf0", "dial||tTr|711") in new stack<br>
-- Executing [s@macro-dial:1] GotoIf("SIP/774-093cfaf0", "1?dial") in new stack<br> -- Goto (macro-dial,s,3)<br> -- Executing [s@macro-dial:3] AGI("SIP/774-093cfaf0", "dialparties.agi") in new stack<br>
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi<br> dialparties.agi: Starting New Dialparties.agi<br> == Parsing '/etc/asterisk/manager.conf': Found<br> == Manager 'admin' logged on from <a href="http://127.0.0.1">127.0.0.1</a><br>
dialparties.agi: Caller ID name is 'comp josimar' number is '774'<br> dialparties.agi: Methodology of ring is 'none'<br> -- dialparties.agi: Added extension 711 to extension map<br> -- dialparties.agi: Extension 711 cf is disabled<br>
-- dialparties.agi: Extension 711 do not disturb is disabled<br> -- dialparties.agi: dbset CALLTRACE/711 to 774<br> -- dialparties.agi: Filtered ARG3: 711<br> == Manager 'admin' logged off from <a href="http://127.0.0.1">127.0.0.1</a><br>
-- AGI Script dialparties.agi completed, returning 0<br> -- Executing [s@macro-dial:7] Dial("SIP/774-093cfaf0", "SIP/711||tTr") in new stack<br> -- Called 711<br> -- SIP/711-093cc3f8 is ringing<br>
== Connect attempt from '<a href="http://127.0.0.1">127.0.0.1</a>' unable to authenticate<br> -- SIP/711-093cc3f8 answered SIP/774-093cfaf0<br> == Parsing '/etc/asterisk/manager.conf': Found<br> == Connect attempt from '<a href="http://127.0.0.1">127.0.0.1</a>' unable to authenticate<br>
-- Started music on hold, class 'default', on SIP/774-093cfaf0<br> -- <SIP/711-093cc3f8> Playing 'pbx-transfer' (language 'pt_BR')<br> -- Stopped music on hold on SIP/774-093cfaf0<br>
== Channel 'SIP/774-093cfaf0' jumping out of macro 'dial'<br> == Channel 'SIP/774-093cfaf0' jumping out of macro 'exten-vm'<br> -- Executing [7777@from-internal-xfer:1] Goto("SIP/774-093cfaf0", "from-pstn|s|1") in new stack<br>
-- Goto (from-pstn,s,1)<br> -- Executing [s@from-pstn:1] NoOp("SIP/774-093cfaf0", "No DID or CID Match") in new stack<br> -- Executing [s@from-pstn:2] Answer("SIP/774-093cfaf0", "") in new stack<br>
-- Executing [s@from-pstn:3] Wait("SIP/774-093cfaf0", "2") in new stack<br> -- Executing [s@from-pstn:4] Playback("SIP/774-093cfaf0", "ss-noservice") in new stack<br> -- <SIP/774-093cfaf0> Playing 'ss-noservice' (language 'pt_BR')<br>
== Parsing '/etc/asterisk/manager.conf': Found<br> == Connect attempt from '<a href="http://127.0.0.1">127.0.0.1</a>' unable to authenticate<br> -- Executing [s@from-pstn:5] SayAlpha("SIP/774-093cfaf0", "") in new stack<br>
== Auto fallthrough, channel 'SIP/774-093cfaf0' status is 'ANSWER'<br><br><br><br><br><br><br clear="all"><br>-- <br>_________________________________________<br>Josimar B. S.