|
Thanks, I will give it a shot tonight.
--
--
Steven
in
the dialplan, before dialing to your legacy pbx, do
a:
Set(CALLERID(name)=)
to "blank" the CID name.
2006/5/15, Steven < [EMAIL PROTECTED]>:
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"
<[EMAIL PROTECTED]>
wrote in message news:[EMAIL PROTECTED]
> 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" <[EMAIL PROTECTED]> wrote
in message news:[EMAIL PROTECTED]
>> 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" <[EMAIL PROTECTED]> wrote
in message news:[EMAIL PROTECTED]
>>>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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