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 >>> >>> >>> >>> >>> _______________________________________________ >>> --Bandwidth and Colocation provided by Easynews.com -- >>> >>> Asterisk-Users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
