search for: prohib_not_screened

Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI> == Using SIP RTP CoS mark 5 -- Executing [100 at sip:1] AGI("SIP/userid-0000001e", "test.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script Executing Application: (SetCallerPr...

Hello, on my ISDN phone I can configure that on the next outgoing call, my telephone number should not be transmitted, instead it should be UNKNOWN. How can I configure Asterisk to do the same? Is this a feature/parameter of the driver (chan_capi) that I'm using? BTW: I'm using ISDN and Deutsche Telekom, if the provider makes any difference. Thanks for your help, Stefan --

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

...*the u() value being dynamically taken from the channel itself. On pbx_b, I have a simply verbose line like this: exten => 5555555555,1,Verbose(1,Presence information : ${CALLERID(num-pres)} - ${CALLERID(name-pres)} - ${CALLERPRES()}) Here is my experience with this: whenever "prohib_not_screened" (tested via a cell phone with hidden caller id info) is sent in the u() value of the Dial application, pbx_b always gets "allowed_not_screened" as presence state. Short version: the callerid presence seems lost on the SIP link. The callerid info isn't, name and number are fin...

...xxx,3,Set(CLI=${CUT(cut1,>,1)}) exten => _00339xxxxxxxx,4,Set(toexten=${CUT(CLI,@,1)}) exten => _00339xxxxxxxx,5,Noop(ORIGINAL NUMBER : [ ${toexten} ]) exten => _00339xxxxxxxx,6,AGI(Ddi-Network.agi,${toexten}) exten => _00339xxxxxxxx,7,Set(CALLERPRES()=prohib_not_screened) exten => _00339xxxxxxxx,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt) exten => _00339xxxxxxxx,9,Hangup and i have in sip.conf: [MyOperator] type=peer host=host-of-my-operator qualify=yes dtmf=rfc2833 nat=no canreinvite=no canredirect=yes insecure=port,invite dtmfmode=rfc2833...

...CID_NAME}) exten = s,n,NoOp(CID_NUMBER: ${CID_NUMBER}) exten = s,n,NoOp(CID_CLIR : ${CID_CLIR}) exten = s,n,NoOp(TRUNK : ${TRUNK}) exten = s,n,Set(CALLERID(name)=${CID_NAME}) exten = s,n,Set(CALLERID(num)=${CID_NUMBER}) exten = s,n,Set(PRESENTATION=${IF($["${CID_CLIR}"="1"]?prohib_not_screened:allowed_not_screened)}) exten = s,n,SetCallerPres(${PRESENTATION}) exten = s,n,GotoIf(${ISNULL(${TRUNK})}?s-CONGESTION,1) exten = s,n,Dial(${TRUNK}/${ARG1}${TRUNKOPTIONS}||gTW) ;Ring the interface exten = s,n,NoOp(DIALSTATUS = ${DIALSTATUS}) exten = s,n,Goto(s-${DIALSTATUS},1) ;Jump based on stat...

Hi; How can a user block his caller-id in Astersik? Regards Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041229/07ecf20f/attachment.htm

Hi, how can I setup asterisk to use the number presentation bits on the isdn side to suppress the number presentation? We need to transmit the subscriber number for billing purposes via ISDN whether or not the user wants to hide his/her number. Is there any way to do this? Deti

I've just upgraded to the latest CVS head, and my outbound calls stopped working. I traced it back to the line exten => s,9,CallingPres(${ARG2}) It seems as if this application is now missing. I tracked back the changes and found in 1.415 of chan_zap.c the code was removed because it was "duplicated". However, it does not exist anywhere ! Am I being stupid, missed

Hi all, on sourceforge.net I added the fixup release 0.5.4 of chan_capi-cm driver. The changes from 0.5.3 to 0.5.4 are: - fixed 'group' setting according to Asterisk defaults. - use SetCallerPres(prohib_not_screened) instead of CallingPres(32) for CLIR. - full CallingPres support added. - use mutex when debug/verbose messages are printed. - set dnid on incoming call. - catch errors in wrong dialstring. - set correct DIALSTATUS and HANGUPCAUSE. - set PROGRESS and PROCEEDING when the network signals them. - incr...

...d presentations are: allowed_not_screened : Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen : Presentation Prohibited, Passed Screen prohib_failed_screen : Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Un...

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

...CID_NAME}) exten = s,n,NoOp(CID_NUMBER: ${CID_NUMBER}) exten = s,n,NoOp(CID_CLIR : ${CID_CLIR}) exten = s,n,NoOp(TRUNK : ${TRUNK}) exten = s,n,Set(CALLERID(name)=${CID_NAME}) exten = s,n,Set(CALLERID(num)=${CID_NUMBER}) exten = s,n,Set(PRESENTATION=${IF($["${CID_CLIR}"="1"]?prohib_not_screened:allowed_not_screened)}) exten = s,n,SetCallerPres(${PRESENTATION}) exten = s,n,GotoIf(${ISNULL(${TRUNK})}?s-CONGESTION,1) exten = s,n,Dial(${TRUNK}/${ARG1}${TRUNKOPTIONS}||gTW) ;Ring the interface exten = s,n,NoOp(DIALSTATUS = ${DIALSTATUS}) exten = s,n,Goto(s-${DIALSTATUS},1) ;Jump based on stat...