Re: [asterisk-users] CDR gets lost

Yes - update your my.conf to increase the timeouts by a large amount, then restart mysql daemon. Here's some details: https://telium.io/en/topic/mysql-server-has-gone-away/ From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Federico Sent: Tuesday,

Re: [asterisk-users] Question about Sip Trunks who support Stir Shaken

Check out Twilio From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Federico Sent: Thursday, August 17, 2023 11:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Question about Sip Trunks who support Stir Shaken

Re: [asterisk-users] Subscribing to events on AMI login

: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua C. Colp Sent: Tuesday, August 8, 2023 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Subscribing to events on AMI login On Tue, Aug 8, 2023 at 12:44 PM TTT

[asterisk-users] Subscribing to events on AMI login

I'm looking at an old app I wrote that upon AMI login would subscribe to events as follows: Action: Login ActionID: myid Username: myun Secret: mypw Events: call, system, security I noticed this old code isn't working, and I *think* that the events parameter of login has been

[asterisk-users] Get manager user info after AMI authentication

Is there an AMI command/action which reports back the username used to authenticate (to the AMI), and the permissions in effect for that user/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

[asterisk-users] Manager permissions for CoreSettings command

I want to use CoreSettings via the AMI. I checked the documentation for the action (command) and it doesn’t list any required permissions: https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_CoreSettings I tried using the CLI “manager show command coresettings” and it

Re: [asterisk-users] AMI versions

-users] AMI versions On Tue, Jul 11, 2023 at 3:40 PM Joshua C. Colp mailto:jc...@sangoma.com> > wrote: On Tue, Jul 11, 2023 at 3:38 PM TTT mailto:li...@telium.io> > wrote: That answers part two…but is there any mapping of AMI version to Asterisk versions? No, there is not.

Re: [asterisk-users] AMI versions

versions https://docs.asterisk.org/latest/Configuration/Interfaces/Asterisk-Manager-Interface-AMI/Asterisk-Manager-Interface-AMI-Changes/ On Tue, Jul 11, 2023 at 11:54 AM, TTT mailto:On%20Tue,%20Jul%2011,%202023%20at%2011:54%20AM,%20TTT%20%3c%3ca%20href=> > wrote: Is there a we

[asterisk-users] AMI versions

Is there a web page that lists the AMI versions mapped to Asterisk versions? I noticed that the AMI version increased quickly to 9.0.0. Will the AMI version increase with each Asterisk version increase in the future? Thanks Brian --

Re: [asterisk-users] Getvar of CHANNEL not working for a couple of items

Sent: Wednesday, July 5, 2023 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Getvar of CHANNEL not working for a couple of items On Wed, Jul 5, 2023 at 12:50 PM TTT mailto:li...@telium.io> > wrote: Channel A: "1688509741.

Re: [asterisk-users] Getvar of CHANNEL not working for a couple of items

Channel A: "1688509741.112" , name: "PJSIP/111-0064" , is originator: Y , call-Id: "u.l6kcou25ca...@mydomain.com " , local_uri: "mailto:sip%3a...@mydomain.com> ;user=phone>" , local_tag: "1734d973-c4da-4ae8-a37d-5f7065f1fe54" , local_addr:

Re: [asterisk-users] Getvar of CHANNEL not working for a couple of items

. In regards to the reversed URI's, am I reading them wrong? Should A's local URI be how to reach A, and A's remote URI be how to reach B ? The missing local and remote addresses for B is just strange (mentioned in previous email) Thanks Brian From: asterisk-users [mailto:as

[asterisk-users] Getvar of CHANNEL not working for a couple of items

The following AMI command works well for all of the information I want: action: Getvar actionid: act1 channel: PJSIP/Twilio-NA-W-3-In-0028 Variable: CHANNEL(pjsip,) Where can be one of the many available items. However, when I create a call from A to B, all of the items return

Re: [asterisk-users] Get channel variables via ARI/AMI

The uppercase command made a difference. I now get a call-id as show below. However, does the call-id look valid? The @0.0.0.0 seems strange. action: Getvar actionid: act1 channel: PJSIP/Twilio-NA-W-3-In-0028 Variable: CHANNEL(pjsip,call-id) Response: Success ActionID: act1

Re: [asterisk-users] Get channel variables via ARI/AMI

>> 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

Re: [asterisk-users] Get channel variables via ARI/AMI

>> 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 “action”, but rather a dialplan

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 U

[asterisk-users] SetCallerPres command gone

) 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

[asterisk-users] AGI script commands

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

Re: [asterisk-users] Get channel variables via ARI/AMI

-users] Get channel variables via ARI/AMI On 6/26/23 5:19 PM, Jeff LaCoursiere wrote: On 6/26/23 9:00 AM, Joshua C. Colp wrote: On Mon, Jun 26, 2023 at 10:57 AM TTT mailto:li...@telium.io> > wrote: I am connecting to the ARI with subscribe all, so I can see channels being created. I no

Re: [asterisk-users] Get channel variables via ARI/AMI

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Monday, June 26, 2023 6:20 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Get channel variables via ARI/AMI On 6/26/23 9:00 AM, Joshua C. Colp wrote: On Mon, Jun 26, 2023 at 10:57 AM TTT mailto:li

Re: [asterisk-users] Get channel variables via ARI/AMI

, June 26, 2023 3:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Get channel variables via ARI/AMI On Mon, Jun 26, 2023 at 4:35 PM TTT mailto:li...@telium.io> > wrote: I think that’s getting me close. I’m trying to get (or recreate) th

Re: [asterisk-users] Get channel variables via ARI/AMI

Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Get channel variables via ARI/AMI On Mon, Jun 26, 2023 at 4:04 PM TTT mailto:li...@telium.io> > wrote: It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the entire SIP header for a channel. I als

Re: [asterisk-users] Get channel variables via ARI/AMI

, 2023 10:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Get channel variables via ARI/AMI On Mon, Jun 26, 2023 at 10:57 AM TTT mailto:li...@telium.io> > wrote: I am connecting to the ARI with subscribe all, so I can see channels being creat

[asterisk-users] Get channel variables via ARI/AMI

I am connecting to the ARI with subscribe all, so I can see channels being created. I now want to extract a variety of header variables (at the moment the from and to tag). I tried to read them from the ARI but Asterisk refuses since the channel is not in a stasis app. Is there a way to

[asterisk-users] Why is WebRTC treated differently from regular SIP in Asterisk

I'm learning about WebRTC clients, and am wondering why Asterisk treats them differently from any other SIP client. The media (RTP) should be no different, so the only difference should be on the signaling side. I noticed that the Asterisk wiki mentions the need for

[asterisk-users] WebRTC signaling

I'm looking at using Asterisk 20 with WebRTC clients (sipjs). I know the media runs over TCP, but what about the signaling? I read something about signaling over UDP was proposed as part of a webrtc standard, but can't find if that was ever ratified or if Asterisk can even use UDP for the

Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

...@lists.digium.com] On Behalf Of TTT Sent: Wednesday, June 21, 2023 2:58 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE I tried that (only needed to add rewrite_contact=yes) but it didn't help. BTW

Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

Asterisk to do that. -Original Message- From: Eric Wieling [mailto:ewiel...@nyigc.com] Sent: Wednesday, June 21, 2023 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion ; TTT Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE type

Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

@lists.digium.com Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE You need to put your external IP in the transport configuration: external_media_address=X.X.X.X external_signaling_address=X.X.X.X external_signaling_port=5060 On 21/06/23 12:36, TTT wrote

[asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

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" ;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4 The IP address above is an

Re: [asterisk-users] PJSIP not performing outbound authentication

m.com] On Behalf Of TTT Sent: Wednesday, June 21, 2023 1:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] PJSIP not performing outbound authentication I didn't use pjsip_wizard, I'm kind of crafting this by hand as I learn. I actually have

Re: [asterisk-users] PJSIP not performing outbound authentication

ers] PJSIP not performing outbound authentication On Wed, Jun 21, 2023 at 05:19:11PM +, 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 > a

Re: [asterisk-users] Multiple phones on same PJSIP account

: [asterisk-users] Multiple phones on same PJSIP account On Wednesday 21 June 2023 at 17:52:16, TTT wrote: > Ok I've got multiple phone sets registered with the same extension/secret. > > However, this causes a strange problem. If I have 3 phone sets > registered on extension 123, and

Re: [asterisk-users] PJSIP not performing outbound authentication

] On Behalf Of Carlos Chavez Sent: Wednesday, June 21, 2023 1:26 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PJSIP not performing outbound authentication Dis you set "outbound_auth" in your endpoint configuration to Twilio? On 21/06/23 11:19, TTT wrote: &g

[asterisk-users] PJSIP not performing outbound authentication

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

Re: [asterisk-users] Multiple phones on same PJSIP account

...@lists.digium.com] On Behalf Of Antony Stone Sent: Monday, June 19, 2023 10:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple phones on same PJSIP account On Monday 19 June 2023 at 16:26:05, TTT wrote: > That begs another interest

Re: [asterisk-users] Multiple phones on same PJSIP account

bject: Re: [asterisk-users] Multiple phones on same PJSIP account On Monday 19 June 2023 at 15:09:44, TTT wrote: > I am creating a dialplan where a single user (Alice) has two offices. > Both of her phones should ring if her extension is called. > > I could use a ring group, but I

[asterisk-users] Multiple phones on same PJSIP account

I am creating a dialplan where a single user (Alice) has two offices. Both of her phones should ring if her extension is called. I could use a ring group, but I'm wondering can both phones use the same PJSIP extension account (username/secret)? Thanks Brian (ast newb) --

Re: [asterisk-users] Get SIP Call-ID from ARI

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua C. Colp Sent: Saturday, June 17, 2023 2:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Get SIP Call-ID from ARI On Sat, Jun 17, 2023 at 2:55 PM TTT ma

[asterisk-users] Get SIP Call-ID from ARI

Based on postings it should be possible to get the SIP Call-ID header value from the ARI. At what point is this value available ? As well, how do I retrieve that value - something like GET /channels/{channelId}/pjsip_header?key=Call-Id But that doesn't work. --

[asterisk-users] Add user to conference via ReST/ARI

I'm trying to join a user (at SIP/99) into a conference via REST/ARI. I want the PBX to call the user, and then join him into an existing conference. I have created a conference in FreePBX with number 1234, and name "conf". Conceptually the steps I have so far: 1. Call

[asterisk-users] Event showing who called who

I'm monitoring the ARI, and if extension 1 calls extension 2, it seems that extension 2 enters the bridge first, then extension 1 enters the bridge. Can I safely (always) determine who initiated the call by who is the latest endpoint to enter the bridge? Or is there a better way to know who

Re: [asterisk-users] Listen to ARI events

-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua C. Colp Sent: Wednesday, June 7, 2023 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Listen to ARI events On Wed, Jun 7, 2023 at 12:04 PM TTT mailto:li...@telium.io

Re: [asterisk-users] Listen to ARI events

Of Joshua C. Colp Sent: Wednesday, June 7, 2023 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Listen to ARI events On Wed, Jun 7, 2023 at 10:46 AM TTT mailto:li...@telium.io> > wrote: I’ve reread the documentation a few times, and what

Re: [asterisk-users] Listen to ARI events

in no events being shown, but no error either. Thanks Brian (Ast newbie) From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of TTT Sent: Tuesday, June 6, 2023 8:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users

Re: [asterisk-users] Listen to ARI events

- Non-Commercial Discussion Subject: Re: [asterisk-users] Listen to ARI events On Tue, Jun 6, 2023 at 6:04 PM TTT mailto:li...@telium.io> > wrote: I have the ARI enabled on my Asterisk test box, and want to listen to all events. I can’t find the syntax to do that. Can I only

[asterisk-users] Listen to ARI events

I have the ARI enabled on my Asterisk test box, and want to listen to all events. I can't find the syntax to do that. Can I only listen to events related to a stasis app? I was hoping that a simple wscat command like this would show me all events: wscat -c

[asterisk-users] FW: Ready to throw up my hands in defeat

explains it fairly well. Please reply to the list only (so anyone can jump in and benefit from the discussion too) -Original Message- From: Steve Matzura [mailto:s...@noisynotes.com] Sent: Monday, May 22, 2023 12:15 PM To: TTT Subject: Re: [asterisk-users] Ready to throw up my

Re: [asterisk-users] Ready to throw up my hands in defeat

There are lots of little tweaks/adjustments overlooked in most guides/books. The examples work most of the time, but even a small difference in your environment might break them. I'm pretty sure the list will be able to answer questions to help you figure it out. If you break down your

Re: [asterisk-users] DUNDI anyone?

DUNDI was a great idea and we saw it deployed, but we've watched clients struggle with it. And many eventually give up on it. I don't consider it overly complex, but I suspect it's proper (and safe) configuration is beyond a lot of tel admins. Im curious what the state of dundi deployments

[asterisk-users] Asterisk unable to do DNS lookups

I've noticed on several occasions that if Asterisk starts without a network connection, then even if the network connection is restored, DNS lookups fail. After the connection is restored I can successfully do NSLOOKUPs from the command line, but the IAX2 registration attempts keep failing

[asterisk-users] Name of module to reload for dialplan

I want to reload the dialplan via the AMI, and I found the documentation showing the command: Action: Reload ActionID: Module: And the module names are * cdr * dnsmgr * extconfig * enum * acl * manager * http *

Re: [asterisk-users] externnotify script not running

I can check/do? Thanks again, Mike. On Thursday, March 17, 2022 3:53:46 PM EDT TTT wrote: > Can I suggest you eliminate a few obvious factors... > > Like 1. try a bash script first instead of perl, 2. Have the bash > script just issue a "logger' command > > If that wor

Re: [asterisk-users] externnotify script not running

Can I suggest you eliminate a few obvious factors... Like 1. try a bash script first instead of perl, 2. Have the bash script just issue a "logger' command If that works go back to perl which does the same. If that works, change you perl script to dump all vars to a file..etc. Also, check

Re: [asterisk-users] voicemails and recordings have words repeated

Unless this is a known bug, there are two likely causes…to figure it out: 1, Is the a long delayed echo on the line ? Is it entire message or just part of message with repeated words? All users/callers or just particular ones? 2. Is there a file writing issue? (Is your asterisk running

[asterisk-users] Asterisk start via systemd fails, but its running

I have a fresh Asterisk 18 install on a fresh OS (AWS Linux 2). I used the service file from contrib directory and commented out user and group settings so it runs under root. When I start Asterisk via systemd it waits a long time and then times out. Reporting failure to the command line.