[Asterisk-Users] Re: Odd internal vs. External dialplanissue

[picciuX]

in the dialplan, before dialing to your legacy pbx, do a:

Set(CALLERID(name)=)

to "blank" the CID name.

2006/5/15, Steven <asterisk at tescogroup.com>:
>
> hidecallerid=yes lets me make the calls from asterisk to the panasonic,
> but now I do not have the CID number either.
>
> What is the proper way to configure asterisk to send a callerID number,
> but NOT send any name info???
>
>
>
> zapata.conf:
> context=panasonic
> swichtype=national
> pridialplan=unknown
> prilocaldialplan=unknown
> signalling=pri_net
> usecallerid=yes
> facilityenable=yes
> hidecallerid=yes
> usecallingpres=yes
> echocancel=no
> echocancelwhenbridged=no
> group=2
> channel => 25-47
>
> --
> --
> Steven
>
> http://www.glimasoutheast.org
>
>
>
> "Steven" <asterisk at tescogroup.com> wrote in message
> news:e3o82n$lgh$1 at sea.gmane.org...
> > This fixed the problem.
> >
> > hidecallerid: (Not for FXO trunk lines) For PRI channels, this will stop
> the sending of Caller ID on outgoing calls. For FXS
> > handsets, this will stop Asterisk from sending this channel's Caller ID
> information to the called party when you make a call using
> > this handset. FXS handset users may enable or disable sending of their
> Caller ID for the current call only by lifting the handset
> > and dialing *82 (enable) or *67 (disable); you will then get a
> "dialrecall" tone whereupon you can dial the number of the
> > extension you wish to contact. Default: no.
> >   hidecallerid=yes
> >
> >
> > --
> > --
> > Steven
> >
> > http://www.glimasoutheast.org
> >
> >
> >
> > "Steven" <asterisk at tescogroup.com> wrote in message
> news:e3ngrh$rqv$1 at sea.gmane.org...
> >> OK, I thinks I have narrowed it down.
> >>
> >> Our old Legacy PBX is choking on the callerID name.
> >> I have a separate issue, where I am not getting the CallerID name from
> our Telco yet, so incoming Telco calls forward fine to the
> >> legacy PBX.
> >> Asterisk to Legacy PBX calls transmit the CallerID name and our legacy
> PBX chokes on it.
> >>
> >> I want to leave on CallerID receiving on the Legacy trunk.
> >> I want to leave "asreceived" for callerID so that PSTN to Legacy
> forwards still have the CallerID number in tact.
> >> I want to stop sending the CallerID Name out the Legacy trunk.
> >> How do I go about turning off CallerID name sending on a trunk?
> >>
> >>
> >> Note:
> >> I tried to figure this out, but many of the settings in zapata.confhave very vague descriptions.
> >>
> >> ex:
> >> ; Whether or not to use caller ID
> >> ;usecallerid=yes
> >> Is this inbound, outbound, both? If off, will the ANI be used like
> callerid?
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> --
> >> --
> >> Steven
> >>
> >> http://www.glimasoutheast.org
> >>
> >>
> >>
> >> "Steven" <asterisk at tescogroup.com> wrote in message
> news:e3aunb$6oo$1 at sea.gmane.org...
> >>>I have the following in my extensions.conf
> >>>
> >>> [ext-local]
> >>> exten => _53XX,1,Wait(2)
> >>> exten => _53XX,2,NoOp,Dialing ${EXTEN} from ext-local-custom
> >>> exten => _53XX,3,Macro(dialout-trunk,2,${EXTEN},,)
> >>>
> >>> This is used to match inbound caller-id for my legacy PBX.
> >>> It works fine for inbound calls, but not for internal SIP calls.
> >>>
> >>> If I call from a SIP phone that is also in [ext-local], it looks like
> it is calling, but never connects.
> >>>
> >>> excerpt from log when called from pstn zap PRI:
> >>> Apr 28 14:18:16 VERBOSE[28452] logger.c:     -- Called g2/5386
> >>> Apr 28 14:18:16 DEBUG[28452] channel.c: Set channel Zap/27-1 to read
> format slin
> >>> Apr 28 14:18:16 DEBUG[28452] channel.c: Set channel Zap/2-1 to write
> format slin
> >>> Apr 28 14:18:16 DEBUG[28452] channel.c: Set channel Zap/2-1 to read
> format slin
> >>> Apr 28 14:18:16 DEBUG[28452] channel.c: Set channel Zap/27-1 to write
> format slin
> >>> Apr 28 14:18:16 DEBUG[11073] devicestate.c: Changing state for Zap/27
> - state 2 (In use)
> >>> Apr 28 14:18:16 DEBUG[28457] app_queue.c: Device 'Zap/27' changed to
> state '2' (In use)
> >>> Apr 28 14:18:17 DEBUG[11111] chan_zap.c: Enabled echo cancellation on
> channel 27
> >>> Apr 28 14:18:17 DEBUG[11073] channel.c: Avoiding initial deadlock for
> 'Zap/27-1'
> >>> Apr 28 14:18:17 VERBOSE[28452] logger.c:     -- Zap/27-1 is ringing
> >>>
> >>> excerpt from log when called from internal SIP extension:
> >>> Apr 28 14:18:25 VERBOSE[28477] logger.c:     -- Called g2/5386
> >>> Apr 28 14:18:25 DEBUG[28477] channel.c: Set channel Zap/27-1 to read
> format ulaw
> >>> Apr 28 14:18:25 DEBUG[28477] channel.c: Set channel SIP/5665-e60f to
> write format ulaw
> >>> Apr 28 14:18:25 DEBUG[28477] channel.c: Set channel SIP/5665-e60f to
> read format ulaw
> >>> Apr 28 14:18:25 DEBUG[28477] channel.c: Set channel Zap/27-1 to write
> format ulaw
> >>> Apr 28 14:18:25 DEBUG[28482] app_queue.c: Device 'Zap/27' changed to
> state '2' (In use)
> >>> Apr 28 14:18:25 DEBUG[28477] rtp.c: Ooh, format changed from unknown
> to ulaw
> >>>
> >>> I never get a ringing log entry if dialed from SIP.
> >>> This SIP phone can call other extensions in asterisk as well as native
> (voicemail) and PSTN calls out ZAP/g0.
> >>>
> >>> I have tried various dial strings ( like the Dial command instead of
> the macro) and they all work for incoming PSTN calls and
> >>> not
> >>> for SIP.
> >>>
> >>> I am at a loss where to find the problem.
> >>>
> >>> Please advise.
> >>>
> >>>
> >>> --
> >>> --
> >>> Steven
> >>>
> >>>
> >>>
> >>>
> >>> _______________________________________________
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> >>>
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> >>>
> >>
> >>
> >>
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> >>
> >
> >
> >
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> >
>
>
>
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