search for: callerpr

Good afternoon list, I'm having a problem using the function CALLERPRES() when connection to a Queue(). When I call an extension, before the Dial (), I select the function CALLERPRES () as "unavailable" to link the extension comes as anonymous. But if I call a queue before the Queue (), I select the function CALLERPRES() as "unavailable", but the...

hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of f...

I have been fighting with the ability to set the caller ID when I make outbound calls via a PRI line as well as via my SIP provider. The more I play around the less I understand. There is a setting in chan_dahdi.conf that seems to say do not pay attention to the CALLERPRES value and just allow the ID to be set. This setting is usecallingpres. If this is set to yes then the value of CALLERPRES controls the sending of the caller ID that might be set with Set(CALLERID(num)=${MyCallerID}). With a setting of yes I had to Set(CALLERPRES()=allowed_not_screened) before the...

...tensions, all are same "design": ; Destination: Gambia Type: Fixe exten => _00220X.,1,Set(CDR(CodeCom)=BUS-GMB) exten => _00220X.,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) exten => _00220X.,3,Set(CALLERID(all)=${NUMID}) exten => _00220X.,4,Set(CALLERPRES()=${CALLPRES}) exten => _00220X.,5,Dial(SIP/Trunk-Telco/${EXTEN:2},180,rt) exten => _00220X.,6,Hangup ; Destination: Libya Type: Fixe exten => _00218X.,1,Set(CDR(CodeCom)=BUS-LBY) exten => _00218X.,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) exte...

I should have included the debug output: <PJSIP/Twilio-NA-W-3-In-00000006>AGI Rx << CALLERPRES(allowed) <PJSIP/Twilio-NA-W-3-In-00000006>AGI Tx >> 510 Invalid or unknown command -----Original Message----- From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of TTT Sent: Saturday, July 1, 2023 11:37 AM To: 'Asterisk Users Mailing List - Non-Co...

...0 RetryTime: 0 ; 2nd CID SetVar: callid2=0036204313763 SetVar: azon2=masodik hivas azonosito { V1pr: ehehhe } Context: CustomCallOut-2 ; 2nd phone num Extension: 003617654321 The contexts: [CustomCallOut-1] ; set custom CDR exten => _0X.,1,Set(CDR(azonosito)=${azon1}) exten => _0X.,n,Set(CALLERPRES()=allowed) exten => _0X.,n,Set(CALLERID(number)=<${callid1}>) exten => _0X.,n,Set(KEEPCID=TRUE) ; pass the call to internal routing include => from-internal [CustomCallOut-2] exten => _0X.,1,Wait(1) ; set custom CDR exten => _0X.,2,Set(CDR(azonosito)=${azon2}) exten => _0...

How to implement some of the function into asterisk like *67 "call number blocking" or *57 "call trace" ? I'm connecting to sipura SPA3K outside line by dialing 9+number. Is there a way to get outside dial tone in SPA3K PSTN-Line by dialing "9"? How to program the extension? -- #Joseph

Hi All, In my dialplan and standard asterisk CLI logging i see that i am able to restrict the callerid when dialing out with asterisk. however, on the receiving phone, the callerid is still displayed. When i increment the logging of the pri with "pri set debug on span 1" on the CLI i also get the lower level debugging info from the pri.

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 AGI debug command worked well, and I found the offending command: SetCallerPres(allowed) That worked in Asterisk 13, but from my google searching it looks like this command has disappeared in Asterisk 20 (actually everything after ver 13). I thought it was replaced with CALLERPRES(allowed) but this generated an error too in Asterisk 20. Is there a replacement command? -...

...; _00339xxxxxxxx,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,...

...Netherlands. Can anyone help me sorting out this issue?? Thanks in advance! -- Executing [s at macro-transfer:25] NoOp("SIP/joostkuif-00000003", "gehe= im") in new stack -- Executing [s at macro-transfer:26] NoOp("SIP/joostkuif-00000003", "voor= de SET CALLERPRES() =3D allowed_not_screened") in new stack -- Executing [s at macro-transfer:27] NoOp("SIP/joostkuif-00000003", "CALL= INGPRES =3D 0") in new stack -- Executing [s at macro-transfer:28] Set("SIP/joostkuif-00000003", "CALLE= RPRES()=3Dprohib") in...

...Netherlands. Can anyone help me sorting out this issue?? Thanks in advance! -- Executing [s at macro-transfer:25] NoOp("SIP/joostkuif-00000003", "gehe= im") in new stack -- Executing [s at macro-transfer:26] NoOp("SIP/joostkuif-00000003", "voor= de SET CALLERPRES() =3D allowed_not_screened") in new stack -- Executing [s at macro-transfer:27] NoOp("SIP/joostkuif-00000003", "CALL= INGPRES =3D 0") in new stack -- Executing [s at macro-transfer:28] Set("SIP/joostkuif-00000003", "CALLE= RPRES()=3Dprohib") in...

...ent Proxy Channel) == Registered custom function 'EXTENSION_STATE' func_extstate.so => (Gets an extension's state in the dialplan) == Registered application 'DAHDIBarge' app_dahdibarge.so => (Barge in on DAHDI channel application) == Registered custom function 'CALLERPRES' == Registered custom function 'CALLERID' func_callerid.so => (Caller ID related dialplan functions) [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:760 load_module: G.729A transcoding module version 1.6.0_3.1.4, Copyright (C) 1999-2009 Digium, Inc. [2014-02-12 22:40:07] NOTI...

2018-06-05 15:27 GMT+02:00 George Joseph <gjoseph at digium.com>: Thank you very much, George for replying. > > > On Tue, Jun 5, 2018 at 3:35 AM Olivier <oza.4h07 at gmail.com> wrote: > >> Hi, >> >> After a long discussion with a friend, I would like to ask here: >> >> 1.According SIP RFCs, is possible/recommended to have different values in

...al(SIP/pbx_b/5555555555,,f(${CALLERID(all)})u(${CALLERID(pres)}))) *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...

How to make outgoing calls from DID and what is theway to get incoming calls from DID. -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110526/3f19091d/attachment.htm>

Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

...2:8:37 Author : 'markster' State : 'Exp' Lines : +2 0 Description : Remove accidental libpri dependency (bug #3714) ---------------------------- Revision : 1.415 Date : 2005/3/9 6:42:56 Author : 'markster' State : 'Exp' Lines : +0 -16 Description : Remove duplicate CallerPres application cvs diff -r 1.414 -r 1.415 chan_zap.c (in directory C:\temp\asterisk\asterisk\channels\) Index: chan_zap.c =================================================================== RCS file: /usr/cvsroot/asterisk/channels/chan_zap.c,v retrieving revision 1.414 retrieving revision 1.415 di...

Hi all, I have a problem setting the MSN for outgoing calls. I'm using a HFC PCI card together with zaptel and bristuff. All my outgoing calls are using the same (first|default|main) MSN. In my zapata.conf I tried different values for pridialplan, prilocaldialplan, nationalprefix, etc but without any success. In my extension.conf I'm setting the MSN with: exten =>