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<DIV><FONT face=Arial size=2>Thanks, I will give it a shot tonight.</FONT></DIV>
<DIV><BR>-- <BR>-- <BR>Steven</DIV>
<DIV> </DIV>
<DIV><A
href="http://www.glimasoutheast.org">http://www.glimasoutheast.org</A></DIV>
<DIV> </DIV>
<DIV><BR> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"picciuX" <<A
href="mailto:matteo@picciux.it">matteo@picciux.it</A>> wrote in message <A
href="news:c41ce8440605150848i1fa5ac07o1c61dde0ed409a1@mail.gmail.com">news:c41ce8440605150848i1fa5ac07o1c61dde0ed409a1@mail.gmail.com</A>...</DIV>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="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">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>>>>
<BR>>>><BR>>>><BR>>>>
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