in the dialplan, before dialing to your legacy pbx, do a:<br><br>Set(CALLERID(name)=)<br><br>to "blank" the CID name.<br><br><div><span class="gmail_quote">2006/5/15, Steven <<a href="mailto:asterisk@tescogroup.com">
asterisk@tescogroup.com</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">hidecallerid=yes lets me make the calls from asterisk to the panasonic, but now I do not have the CID number either.
<br><br>What is the proper way to configure asterisk to send a callerID number, but NOT send any name info???<br><br><br><br>zapata.conf:<br>context=panasonic<br>swichtype=national<br>pridialplan=unknown<br>prilocaldialplan=unknown
<br>signalling=pri_net<br>usecallerid=yes<br>facilityenable=yes<br>hidecallerid=yes<br>usecallingpres=yes<br>echocancel=no<br>echocancelwhenbridged=no<br>group=2<br>channel => 25-47<br><br>--<br>--<br>Steven<br><br><a href="http://www.glimasoutheast.org">
http://www.glimasoutheast.org</a><br><br><br><br>"Steven" <<a href="mailto:asterisk@tescogroup.com">asterisk@tescogroup.com</a>> wrote in message news:e3o82n$lgh$1@sea.gmane.org...<br>> This fixed the problem.
<br>><br>> hidecallerid: (Not for FXO trunk lines) For PRI channels, this will stop the sending of Caller ID on outgoing calls. For FXS<br>> handsets, this will stop Asterisk from sending this channel's Caller ID information to the called party when you make a call using
<br>> this handset. FXS handset users may enable or disable sending of their Caller ID for the current call only by lifting the handset<br>> and dialing *82 (enable) or *67 (disable); you will then get a "dialrecall" tone whereupon you can dial the number of the
<br>> extension you wish to contact. Default: no.<br>> hidecallerid=yes<br>><br>><br>> --<br>> --<br>> Steven<br>><br>> <a href="http://www.glimasoutheast.org">http://www.glimasoutheast.org</a>
<br>><br>><br>><br>> "Steven" <<a href="mailto:asterisk@tescogroup.com">asterisk@tescogroup.com</a>> wrote in message news:e3ngrh$rqv$1@sea.gmane.org...<br>>> OK, I thinks I have narrowed it down.
<br>>><br>>> Our old Legacy PBX is choking on the callerID name.<br>>> 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<br>
>> legacy PBX.<br>>> Asterisk to Legacy PBX calls transmit the CallerID name and our legacy PBX chokes on it.<br>>><br>>> I want to leave on CallerID receiving on the Legacy trunk.<br>>> I want to leave "asreceived" for callerID so that PSTN to Legacy forwards still have the CallerID number in tact.
<br>>> I want to stop sending the CallerID Name out the Legacy trunk.<br>>> How do I go about turning off CallerID name sending on a trunk?<br>>><br>>><br>>> Note:<br>>> I tried to figure this out, but many of the settings in
zapata.conf have very vague descriptions.<br>>><br>>> ex:<br>>> ; Whether or not to use caller ID<br>>> ;usecallerid=yes<br>>> Is this inbound, outbound, both? If off, will the ANI be used like callerid?
<br>>><br>>><br>>><br>>><br>>><br>>><br>>><br>>> --<br>>> --<br>>> Steven<br>>><br>>> <a href="http://www.glimasoutheast.org">http://www.glimasoutheast.org
</a><br>>><br>>><br>>><br>>> "Steven" <<a href="mailto:asterisk@tescogroup.com">asterisk@tescogroup.com</a>> wrote in message news:e3aunb$6oo$1@sea.gmane.org...<br>>>>I have the following in my
extensions.conf<br>>>><br>>>> [ext-local]<br>>>> exten => _53XX,1,Wait(2)<br>>>> exten => _53XX,2,NoOp,Dialing ${EXTEN} from ext-local-custom<br>>>> exten => _53XX,3,Macro(dialout-trunk,2,${EXTEN},,)
<br>>>><br>>>> This is used to match inbound caller-id for my legacy PBX.<br>>>> It works fine for inbound calls, but not for internal SIP calls.<br>>>><br>>>> If I call from a SIP phone that is also in [ext-local], it looks like it is calling, but never connects.
<br>>>><br>>>> excerpt from log when called from pstn zap PRI:<br>>>> Apr 28 14:18:16 VERBOSE[28452] logger.c: -- Called g2/5386<br>>>> Apr 28 14:18:16 DEBUG[28452] channel.c: Set channel Zap/27-1 to read format slin
<br>>>> Apr 28 14:18:16 DEBUG[28452] channel.c: Set channel Zap/2-1 to write format slin<br>>>> Apr 28 14:18:16 DEBUG[28452] channel.c: Set channel Zap/2-1 to read format slin<br>>>> Apr 28 14:18:16 DEBUG[28452]
channel.c: Set channel Zap/27-1 to write format slin<br>>>> Apr 28 14:18:16 DEBUG[11073] devicestate.c: Changing state for Zap/27 - state 2 (In use)<br>>>> Apr 28 14:18:16 DEBUG[28457] app_queue.c: Device 'Zap/27' changed to state '2' (In use)
<br>>>> Apr 28 14:18:17 DEBUG[11111] chan_zap.c: Enabled echo cancellation on channel 27<br>>>> Apr 28 14:18:17 DEBUG[11073] channel.c: Avoiding initial deadlock for 'Zap/27-1'<br>>>> Apr 28 14:18:17 VERBOSE[28452]
logger.c: -- Zap/27-1 is ringing<br>>>><br>>>> excerpt from log when called from internal SIP extension:<br>>>> Apr 28 14:18:25 VERBOSE[28477] logger.c: -- Called g2/5386<br>>>> Apr 28 14:18:25 DEBUG[28477]
channel.c: Set channel Zap/27-1 to read format ulaw<br>>>> Apr 28 14:18:25 DEBUG[28477] channel.c: Set channel SIP/5665-e60f to write format ulaw<br>>>> Apr 28 14:18:25 DEBUG[28477] channel.c: Set channel SIP/5665-e60f to read format ulaw
<br>>>> Apr 28 14:18:25 DEBUG[28477] channel.c: Set channel Zap/27-1 to write format ulaw<br>>>> Apr 28 14:18:25 DEBUG[28482] app_queue.c: Device 'Zap/27' changed to state '2' (In use)<br>>>> Apr 28 14:18:25 DEBUG[28477]
rtp.c: Ooh, format changed from unknown to ulaw<br>>>><br>>>> I never get a ringing log entry if dialed from SIP.<br>>>> This SIP phone can call other extensions in asterisk as well as native (voicemail) and PSTN calls out ZAP/g0.
<br>>>><br>>>> I have tried various dial strings ( like the Dial command instead of the macro) and they all work for incoming PSTN calls and<br>>>> not<br>>>> for SIP.<br>>>><br>
>>> I am at a loss where to find the problem.<br>>>><br>>>> Please advise.<br>>>><br>>>><br>>>> --<br>>>> --<br>>>> Steven<br>>>><br>>>>
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