similar to: CallerId presence issue

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 --

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, 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]

I am using VoiceOne http://voiceone.it/ as my management interface. I am not 100% sure when it started, but my CDR is now full of "s" as the DST instead of the actual dialed number. As I understand it - it is because it is being recorded in the CDR while in a macro (as below). Is there any work around so that I can record the actual dialed number? [macro-dialout] exten =

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

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>

Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Mon, 13 Jul 2020 15:44:12 -0400, Matthew Fredrickson wrote: > > On Mon, Jul 13, 2020 at 2:34 PM Saint Michael <venefax at gmail.com> wrote: > >> > >> There is a big confusion here about Stir Shaken. It is NOT a provider issue. Un fact, all providers are whasing their hands and modifying their swihtches to pass-through the Signature. They cannot sign the call

Hi i use this into my extension : exten => _00339xxxxxxxx,1,Set(foo=${SIP_HEADER(To)}) exten => _00339xxxxxxxx,2,Set(cut1=${CUT(foo,:,2)}) exten => _00339xxxxxxxx,3,Set(CLI=${CUT(cut1,>,1)}) exten => _00339xxxxxxxx,4,Set(toexten=${CUT(CLI,@,1)}) exten => _00339xxxxxxxx,5,Noop(ORIGINAL NUMBER : [ ${toexten} ]) exten =>

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 -

On 1 July 2016 at 17:41, Joshua Colp <jcolp at digium.com> wrote: > > >> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) >> same => n,Dial(PJSIP/phone123, 30) >> > > Your exten line has no priority, is that how it is in your dialplan? > Actually no, I stole that line from an earlier email to this list. Mine has a priority.

I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place calls inbound, everything works fine. If I place calls outbound, originating from the Asterisk box, everything works fine (I have done this with the use of the .call files). If I setup an extension with the findme-followme feature and have it try to hair-pin a call back out the same trunk to the Avaya, I get a

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, Is it possible to set a variable in dialpan when the someone calls a particular DID number so that i can use that variable for calls coming to that number only. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/25402ade/attachment.htm

Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow.

Dear list, I have the function (as a simple example, which is actually part of a larger function) pres.test<-function(N0=N0, N1=N1) { dt<-5 r<-log(N1/N0)/dt r } which calculates the annual growth rates in a population Where N0 is the population classified into age intervals, say 5 years, at time=1995, and N1 is the population by 5 year age classes at time=2000.

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

List, I want to reshape my data, but I'm not sure how to do it... it might be a simple task, but don't know which package does this. "occ.data" (see below) is how my original data are arranged, and I know that with melt() I can reshape it like "y" (see below). However, I just want to build a matrix like the "y" matrix, but with only 2 dimensions. Something

Hi All, When receiving an invite containing two different caller ID, one in FROM header and the other in "P-Asserted Identity" Header, Which one will be used by the callee ? I couldn't find any RFC specifying this detail. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: