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

Steven asterisk at tescogroup.com
Mon May 15 11:04:57 MST 2006


Nope, that didn't work.

The idea made sense though.

It must be a PRI thing and any CIDName info, even null, makes the Legacy PBX stop responding on that channel.
It doesn't hang-up, by it never reports ringing over the PRI either.

-- 
-- 
Steven

http://www.glimasoutheast.org



  "Steven" <asterisk at tescogroup.com> wrote in message news:e4afhh$ihr$1 at sea.gmane.org...
  Thanks, I will give it a shot tonight.

  -- 
  -- 
  Steven

  http://www.glimasoutheast.org


   
    "picciuX" <matteo at picciux.it> wrote in message news:c41ce8440605150848i1fa5ac07o1c61dde0ed409a1 at mail.gmail.com...
    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.conf have 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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      > Asterisk-Users mailing list 
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