[Asterisk-Users] asterisk to analog pbx

[Julio Saura]

Hi

this is the macro used for that purpose ..

[macro-dialout-trunk]
exten => s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4)        ;check
for CID override for exten
exten => s,2,SetCallerID(${ECID${CALLERIDNUM}})
exten => s,3,Goto(6)
exten => s,4,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6)            ;check
for CID override for trunk
exten => s,5,SetCallerID(${OUTCID_${ARG1}})
exten => s,6,SetGroup(OUT_${ARG1})
exten => s,7,CheckGroup(${OUTMAXCHANS_${ARG1}})
; if we've used up the max channels, continue at 108 (n+101)
exten => s,8,SetVar(DIAL_NUMBER=${ARG2})
exten => s,9,SetVar(DIAL_TRUNK=${ARG1})
exten => s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper
dial string for this trunk
exten => s,11,Dial(${OUT_${ARG1}}/${OUTPREFIX_${ARG1}}${DIAL_NUMBER})
; if dial fails (ie, all channels are busy), continue at 112 (n+101)
;exten => s,11,Dial(Zap/0/${DIAL_NUMBER})

; we should only get here if the call was successful (?)
exten => s,9,Congestion

; exit points for macro
exten => s,108,NoOp(max channels used up)
exten => s,112,NoOp(dial failed)

as u can see is also a dial instruction

the call seems to be done but in fact my analog extension does not
ring :/

any clue?
Thanks again





El mar, 03-05-2005 a las 10:17 -0500, Moises Silva escribió:
> Hi Julio. It would be nice if you show the extensions.conf that
> handles that kind of calls. You can do something like this:
> 
> [macro-analogpbx]
> exten => s,1,Cut(ChannelType=CHANNEL,/,1) //check if the call comes
> from other Zap ch
> exten => s,2,GotoIf($[${ChannelType} = Zap] ? 3 : 6) //If does, go 3, othewise 6
> exten => s,3,Flash() 
> exten => s,4,SendDTMF(${analogprefix}${num}) //send the DTMF for the
> extension dialed
> exten => s,5,Hangup() 
> exten => s,6,Dial(Zap/g${analoggroup}/${analogprefix}${num}) //if the
> call comes from SIP or IAX then execute Dial trough some group in
> zapata
> exten => s,7,Hangup() 
> 
> You can see some variables i just use for administration of my PBX,
> but i hope you understand the concept.
> 
> Good Look
> 
> - moy
> 
> On 5/3/05, Julio Saura <julio.saura at dbs.es> wrote:
> > Hi there
> > 
> > i have an asterisk box running ok, and now i am trying to integrate it
> > with my local analog pbx
> > 
> > So far, i have connected the fxo port of my * to an analog extension
> > port of my analog pbx.
> > 
> > As far as i know, if a call an extension of my analog pbx on a sip phone
> > ( i have done the right dial plan for routing these calls to de zap
> > channel ) the analog pbx extension should ring ...
> > 
> > am i right?
> > 
> > asterisk says the call is done, but the analog extension keeps in
> > silence .. :?
> > 
> > any clue, am i doing something wrong?
> > 
> > Best regards.
> > 
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