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
