Re: [asterisk-users] SetCallerPres command gone
On 7/1/2023 11:40 AM, TTT wrote: > I thought it was replaced with CALLERPRES(allowed) but this generated an > error too in Asterisk 20. From UPGRADE.txt¹: The CALLERPRES() dialplan function is deprecated in favor of CALLERID(num-pres) and CALLERID(name-pres). Kind regards, Sean 1. https://github.com/asterisk/asterisk/blob/b2cdb530dd0619eff9c45271155fa5aedcc0c855/UPGRADE.txt#L1827-L1828 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SetCallerPres command gone
I should have included the debug output:
AGI Rx << CALLERPRES(allowed)
AGI Tx >> 510 Invalid or unknown command
-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of TTT
Sent: Saturday, July 1, 2023 11:37 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
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 CALLERPRES(allowed) but this generated an error
too in Asterisk 20.
Is there a replacement command?
-Original Message-
From: Eric Wieling [mailto:ewiel...@nyigc.com]
Sent: Saturday, July 1, 2023 1:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
; TTT
Subject: Re: [asterisk-users] AGI script commands
You have to read stdin to accept the data Asterisk sends when the AGI
starts before you can send any AGI commands to Asterisk. Also, "agi
set debug on".
On 6/30/23 21:52, TTT wrote:
> I have an AGI script written in PHP that worked great with Asterisk 13.
> I’m porting it to an Asterisk 20 site and have a strange problem. I
> tried running the script from the command line and it works fine; I
> see the script commands written to stdout like
>
> VERBOSE “SmartScreen v1”
>
> But when run from asterisk the CLI shows:
>
> [2023-06-30 15:50:47] VERBOSE[1264031][C-0025] pbx.c: Executing
> [s@function-smartscreen:2] EAGI("PJSIP/Twilio-NA-W-3-In-0068",
> "smartscreen/smartscreen.php,"GEORGE SMITH" <+1234567890>") in new
> stack
>
> [2023-06-30 15:50:47] VERBOSE[1264031][C-0025] res_agi.c: Launched
> AGI Script /var/lib/asterisk/agi-bin/smartscreen/smartscreen.php
>
> [2023-06-30 15:50:48] VERBOSE[1264031][C-0025] res_agi.c:
> AGI Script
> smartscreen/smartscreen.php completed, returning 0
>
> I never see any messages or commands sent from the script to stdout
> (to
> asterisk) Has the way EAGI operates changed? This script doesn’t use
> any AGI libraries…just simply read/write to stdin/stdout.
>
>
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[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 CALLERPRES(allowed) but this generated an error
too in Asterisk 20.
Is there a replacement command?
-Original Message-
From: Eric Wieling [mailto:ewiel...@nyigc.com]
Sent: Saturday, July 1, 2023 1:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
; TTT
Subject: Re: [asterisk-users] AGI script commands
You have to read stdin to accept the data Asterisk sends when the AGI
starts before you can send any AGI commands to Asterisk. Also, "agi
set debug on".
On 6/30/23 21:52, TTT wrote:
> I have an AGI script written in PHP that worked great with Asterisk 13.
> I’m porting it to an Asterisk 20 site and have a strange problem. I
> tried running the script from the command line and it works fine; I
> see the script commands written to stdout like
>
> VERBOSE “SmartScreen v1”
>
> But when run from asterisk the CLI shows:
>
> [2023-06-30 15:50:47] VERBOSE[1264031][C-0025] pbx.c: Executing
> [s@function-smartscreen:2] EAGI("PJSIP/Twilio-NA-W-3-In-0068",
> "smartscreen/smartscreen.php,"GEORGE SMITH" <+1234567890>") in new
> stack
>
> [2023-06-30 15:50:47] VERBOSE[1264031][C-0025] res_agi.c: Launched
> AGI Script /var/lib/asterisk/agi-bin/smartscreen/smartscreen.php
>
> [2023-06-30 15:50:48] VERBOSE[1264031][C-0025] res_agi.c:
> AGI Script
> smartscreen/smartscreen.php completed, returning 0
>
> I never see any messages or commands sent from the script to stdout
> (to
> asterisk) Has the way EAGI operates changed? This script doesn’t use
> any AGI libraries…just simply read/write to stdin/stdout.
>
>
--
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Re: [asterisk-users] SetCallerPres questions
Hi, In 1.8 setting this flag it does not remove any caller id related data, it just sets an information on how to handle the data. If the direct dial target is a phone, it MAY show the callerid anyway, but dialling out on a PSTN/VoIP trunk also sends the callerid. But: If you place an emergency call the emergency service should always be able to see your callerid - even if you set SetCallerPres(Prohib). Max Am 16.05.2013 00:12, schrieb Adam Moffett: > 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. > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SetCallerPres questions
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. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SetCallerPres
On 2 Sep 2008, at 10:36, Steven Howes wrote:
> asterisk-1.4.21.2
> libpri-1.4.7
> zaptel-1.4.11
>
> I might be being a muppet here (not used PRI with Asterisk before) so
> humor me.. I am using SetCallerPres on an outbound call over PRI...
> Console shows:
>
> -- Executing [EMAIL PROTECTED]:8] SetCallerPres("SIP/XXX.XXX.
> 209.243-08b81d68", "prohib_failed_screen") in new stack
>
> Which is what I want.. However pri debug shows:
>
>> Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>> Presentation: Presentation permitted,
> user number passed network screening (1) '1598' ]
>
> Which seems a bit odd to me... Just for completeness I did a
> allowed_passed_screen
>
> -- Executing [EMAIL PROTECTED]:8] SetCallerPres("SIP/XXX.XXX.
> 209.243-b760f2d8", "allowed_passed_screen") in new stack
>
>> Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>> Presentation: Presentation permitted,
> user number passed network screening (1) '159849' ]
>
>
> Again it doesn't appear to be listening to me.. Am I missing something
> obvious or is there actually something wrong here... Can post zap
> configs if needed..
>
> Thanks in advance.
>
> Steve
As originally suspected, i am a muppet... "usecallingpres=yes" is
required...
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[asterisk-users] SetCallerPres
asterisk-1.4.21.2
libpri-1.4.7
zaptel-1.4.11
I might be being a muppet here (not used PRI with Asterisk before) so
humor me.. I am using SetCallerPres on an outbound call over PRI...
Console shows:
-- Executing [EMAIL PROTECTED]:8] SetCallerPres("SIP/XXX.XXX.
209.243-08b81d68", "prohib_failed_screen") in new stack
Which is what I want.. However pri debug shows:
> Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> Presentation: Presentation permitted,
user number passed network screening (1) '1598' ]
Which seems a bit odd to me... Just for completeness I did a
allowed_passed_screen
-- Executing [EMAIL PROTECTED]:8] SetCallerPres("SIP/XXX.XXX.
209.243-b760f2d8", "allowed_passed_screen") in new stack
> Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> Presentation: Presentation permitted,
user number passed network screening (1) '159849' ]
Again it doesn't appear to be listening to me.. Am I missing something
obvious or is there actually something wrong here... Can post zap
configs if needed..
Thanks in advance.
Steve
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Re: [asterisk-users] SetCallerPres and Cisco PRI
PS. Check this out: http://bugs.digium.com/print_bug_page.php?bug_id=2471 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SetCallerPres and Cisco PRI
Hi Peder, You tried blanking the caller ID field and it didn't work? i.e., exten => ...,n,Set(CALLERID(all)=) It worked for me, although my media gateway was not a Cisco one. Whether SetCallerPres() will work depends entirely on what it accomplishes. Does it just alter the cosmetic From: line, and does the Cisco gateway take stock in that? Or does it tack on the draft privacy headers (Remote-Party-ID) and set privacy to on/full? My gut feeling is that SetCallerPres() applies to calls placed directly out of a PRI interface, not SIP, because "presentation" is a term typically applied to caller ID in an ISDN, not a SIP context. It is hard to tell whether this intuition is correct because SetCallerPres() is fundamentally implemented in apps/app_setcallerid.c which calls ast_set_callerid() in main/channel.c and appears to apply to a variety of channel types variously. If this doesn't work, try this: http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header A Cisco MGW should support that just fine. Good luck, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SetCallerPres and Cisco PRI
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 added as teh callerid, so we can't do that. We need to make it anonymous so that it shows as unknown on the other end. Peder ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] setcallerpres not working
Hi I have the following setup phone - mta - asterisk - patton_sn2400 - PRI I am trying to program *67 to block caller id name and number To do this correctly I have to leave the fields callerid name and number unchanged and only set the flag callerpres to restricted The problem seems to be that Asterisk replace the name and number to unknown and then send the call to my Patton box. How can I make this setup work ? Patrick ___ --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
[Asterisk-Users] SetCallerPres
I'm trying to set caller presentation to prohibited and I'm having slight problems doing it. Using a machine that has a Sangoma facing my Telco works but when using an asterisk that talks to the first machine using SIP it does not work. I suspect that SetCallerPres is not transitive, ie it's not communicated between SIP peers but need to be set at the actual machine having the Sangoma card, correct? Anyone have a workaround for this? How should I set callerpres to prohib when doing SIP to SIP calls? Or when calling via SIP and then out on the PRI, how can I set callerpres on the machine originating the call? Thank you Regards, Kristian ___ --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
Re: [Asterisk-Users] SetCallerPres problem
hi, did you solve this problem, which i exactly have? lokotes wrote: Hi, Background: I'm running 2x * boxes. Box A has a registered user which dials a number. The connection is sent to Box B which acts as pstn gateway (sangoma 1xE1 card). Problem: On Box A before executing Dial() command I set SetCallerPres(prohib_no_screened) but despite that Box B sends the connection to pstn with allowed_not_screened flag ? Why is that? When I set SetCallerPres(prohib_no_screened) on Box B it acts properly. But why sending this flag between 2 8 boxes doesn't work for me? Any suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SetCallerPres problem
Hi, Background: I'm running 2x * boxes. Box A has a registered user which dials a number. The connection is sent to Box B which acts as pstn gateway (sangoma 1xE1 card). Problem: On Box A before executing Dial() command I set SetCallerPres(prohib_no_screened) but despite that Box B sends the connection to pstn with allowed_not_screened flag ? Why is that? When I set SetCallerPres(prohib_no_screened) on Box B it acts properly. But why sending this flag between 2 8 boxes doesn't work for me? Any suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
