search for: setcallerpres

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...

Hi, Using * 1.2.5 with a euro_isdn PRI I need to hide the callerID on certain extensions. I have usecallingpres=yes in zapata.conf, and am using SetCallerPres(prohib) in my dialplan prior to the Dial command. No matter what I set SetCallerPres to the CID is still displayed. Is there something else I need to make this work? I can't just set the CallerIDNUM to null, as it is needed for billing purposes. Thanks. -Barry Flanagan

Hello SetCallerPres function seems to be removed from Asterisk 1.4. What function or application replaced it? Bit of a problem if you want to use CLIR on your PRI connections. Jon No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.7.6/815 - Release Date...

Does anybody know if SetCallerPres works on calls via SIP through a Cisco gateway? We are trying to mark outbound calls as anonymous and we set it to prohib, but calls still show outbound callerid. We are SIP from * to the Cisco gateway and then PRI outbound. If we strip the callerid num, then the first number on the PRI gets...

Does SetCallerPres(Prohib) remove the ANI data from a SIP call or does it simply set a flag telling other devices not to display the data? In other words, could another system override that and see the caller ID anyway? The answer may affect how I handle 911 calls, so I'm very curious.

I have an issue where a particular dialplan works but another doesn't. I'm not sure why. To me they look identical and it has me stumped. This works: [to-test] exten => _X., 1, SetCallerPres(allowed) exten => _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb) exten => _X., 3, Ringing exten => _X., 4, Dial(SIP/9330 at a-test,20,ro) exten => _X., 5, GotoIf($["${DIALSTATUS}" = "ANSWER"]?9) exten => _X., 6, GotoIf($["${DIALSTATUS}" = "NOANSWER&q...

...ays ago I found out with help of some of you guys how to set CLIR. (Calling line identification restriction) My first idea was to use the keypad protocol to set the CLIR with dialing *31* before the number but this was not possible. So thanks to Damon Estep I got it to work with executing 'SetCallerPres(prohib)' before the dial command. This works perfectly! But now for my new issue: I want to use the CLIR for automated dial out like described on http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out There is no way (at least I was not able to find it yet) to execute the '...

...and -----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-Commercial Discussion' <asterisk-users at lists.digium.com> Subject: [asterisk-users] SetCallerPres command gone 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 CALLERP...

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? ---...

Hi, how may I define in a callfile the CallerID presentation to be used for the requested call, eg. set it to prohibited? TIA, Bruno -------------- next part -------------- A non-text attachment was scrubbed... Name: Bruno.Voigt.vcf Type: text/x-vcard Size: 270 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050913/fcb5c595/Bruno.Voigt.vcf

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 --

hi all! I'm running an Asterisk-box with bristuff-....RC8n and 2 HFC-S cards. I m located in Vienna/Austria. I have the problem that on outgoing calls i hear my voice as a short echo (about half a second). This occurs not on every call. I tried some changes in my zapata.conf, with rxgain and txgain settings, but to me its hard to find a configuration which is good for every call i make. Is

I am trying to setup one time caller id block on my system(activated when an incoming call matches *811XXXXXXXXXX), and I have had little to no luck. Could you take a look at my context/macro definition and help me figure out what I am missing? Here is my context for my dialplan: include=default plancomment=user-default

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

...and facility IE, which are all elements of the ISDN signalling protocol. I would be very surprised if the same methods used in the US did not work in the Netherlands in place of the keypad protocol. That would be; SetCallerID(<calleridvalue>|a) And SetCallerIDNumber(<caleridvalue>) And SetCallerPres (<presentation>) <- this one is in newer code and allows setting of the presentation flags in the ISDN setup message. Try setting these plags if you are using any type of isdn signalling and see if they are accepted (that is the feature is activated for he call). Here are the values used...

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

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

...ward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no switchtype=5ess context=main signalling=pri_cpe group=1 channel => 1-23 channel => 25-47 exten => 1234,1,Wait,1 exten => 1234,2,Answer exten => 1234,3,SetCallerPres(allowed_passed_screen) exten => 1234,4,SetCIDNum(8881234567) exten => 1234,5,Dial(Zap/g1/18887654321,,,) exten => 1234,6,Hangup Thanks, Shaun Tierney

...d misdn 0.3.1rc27. I use an e1 card with sip clients. My extensions look like this: [E1] <snip>...<snip> exten => 33006733,1,Set(CALLED=${EXTEN}) exten => 33006733,2,Dial(SIP/1@192.168.0.23) exten => 33006733-ANSWER,3,Answer() [SIP] exten => _X.,1,Noop() exten => _X.,2,SetCallerPres(allowed_passed_screen) exten => _X.,3,Dial(mISDN/g:E1/${EXTEN},40) exten => _X.-BUSY,4,Busy(1) But whenever a sip client calls to an exten that is busy through e1 I get busy tones for 10s before I get disconnected. But I want to have it only for 1s. Does anyone know how to fix that? regards...

Hi Chris, Yes the call was from PSTN and your solution worked great! I've read about SetCallerPres earlier but I didn't connect the dots this time. Thanks alot! :) Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Christoph F?rstaller Skickat: den 29 mars 2007 15:29 Till: Asterisk Users Mailing...