similar to: SetCallerPres command gone

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 -

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]

I tried that (only needed to add rewrite_contact=yes) but it didn't help. BTW, the CONTACT: line holds the correct ip! Only the FROM: line holds the wrong (private) IP. I'm still learning SIP...but I assume the FROM should also hold the rewritten public IP. Just don't know how to force Asterisk to do that. -----Original Message----- From: Eric Wieling [mailto:ewieling at

Something perhaps noteworth, since this is a multihomed system I bound the transport to 172.31.253.4:5060 I don't *think* that would cause Asterisk to use that IP in the FROM...at least it shouldn't. -----Original Message----- From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of TTT Sent: Wednesday, June 21, 2023 2:58 PM To: 'Asterisk Users Mailing

type=endpoint rewrite_contact=yes force_rport=yes rtp_symmetric=yes On 6/21/23 14:36, TTT wrote: > I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction: > > From: "MYNAME" <sip:16667778888 at

I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction: From: "MYNAME" <sip:16667778888 at 172.31.253.4>;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4 The IP address above is an internal/non-routable IP, so Twilio is rejecting it. For some

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 Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: [Twilio] type=auth auth_type=userpass password=mysecret username=myun However, my calls using the trunk are rejected with a 403. Using pjsip logging I notice that the outgoing invite does not have an authentication line. Why is Asterisk not sending

On Wed, Jun 21, 2023 at 05:19:11PM +0000, TTT wrote: > I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP > (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: > > [Twilio] > type=auth > auth_type=userpass > password=mysecret > username=myun > > However, my calls using the trunk are rejected with a 403. Using pjsip >

I didn't use pjsip_wizard, I'm kind of crafting this by hand as I learn. I actually have a plain asterisk, and a FreePBX, system to help me learn. I sometimes create something in FreePBX to see what it does to the config files. So that's how I modelled my pjsip.X.conf files If I issue the command "pjsip show endpoint Twilio" it does show that outbound_auth=Twilio Does

    Dis you set "outbound_auth" in your endpoint configuration to Twilio? On 21/06/23 11:19, TTT wrote: > I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP > (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: > > [Twilio] > type=auth > auth_type=userpass > password=mysecret > username=myun > > However, my calls

>> You use the AMI action Getvar[1] which allows channel variables and dialplan functions. >> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar I actually tried that, and although I get “success” I never get useful data. For example: action: Getvar actionid: act1 channel: PJSIP/Twilio-NA-W-2-In-00000025 Variable: channel(pjsip,call-id)

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 full and yes how can I correct this?

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

On Sun, Jul 2, 2023 at 4:18 PM TTT <lists at telium.io> wrote: > >> There are SOME protocol level things accessible using CHANNEL[1] but > that's it. > > >> [1] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL > > > > > > I am trying to use the CHANNEL function listed above from the AMI. Since > it is not an AMI

Does anyone have concern about this? What if Redhat stops giving SRPMS for new releases and updates in public? If you buy a subscription to RHEL AS and they have to give you the SRPMS because of GPL agreement can that person provide those SRPMS to public again because of GPL? They claim they are premium OpenSource company but they are still a company and for them money is bottom line.

On Sun, Jul 2, 2023 at 4:39 PM TTT <lists at telium.io> wrote: > >> You use the AMI action Getvar[1] which allows channel variables and > dialplan functions. > > >> [1] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar > > > > > I actually tried that, and although I get “success” I never get useful > data. For

In an earlier post I made about learning how to use speex someone nicely responded to me with a Speex Wrapper they had written.. however I am having an issue now where all I seem to hear is a ticking noise after encoding and decoding with speex. It is definitely getting proper data, as if I hit the mic somewhat I hear some extra crackling noises. Now, my code was nearly the same... but...

Yes, I have tried that too (i.e, exten => +17775551212,1,Log(WARNING, TWILIO)). It does not work and NO error message in CLI. I have also tried http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ since I first emailed this group, but that does not seem to work either. Here is my log: [Dec 2 15:09:28] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: Call from

Greetings ! My goal is to get Twilio trunking working, and with TLS/SRTP. I see this concerning message in my log: [Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object of type 'endpoint' with id ?twilio' from configuration file ?pjsip.conf? Thus, ?pjsip show endpoints? does not show the endpoint for the Twilio trunk. Hoping for a sanity check of