[Asterisk-Users] Re: Re: Odd internal vs. External dialplanissue
Steven
asterisk at tescogroup.com
Mon May 15 10:57:37 MST 2006
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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