similar to: Does Asterisk cache AstDB?

Hello, On Asterisk 13.19 I'm trying to set astdbdir in /etc/asterisk.conf, but it's not having any effect. For example: # grep astdbdir /etc/asterisk/asterisk.conf astdbdir => /tmp/asterisk /tmp/asterisk exists and is owned by asterisk:asterisk, as the asterisk processing is running as asterisk:asterisk with the config file forced: # ps aux | grep asterisk asterisk 3389 13.1 3.5

It depends a bit on your version of FreePBX, but here's a link to show you how: http://telium.ca/pages/forums/viewtopic.php?f=7&t=19 Hopefully option 1 works for you (quick and easy). If not, you'll have to try option 2. Ignore option 3 since that's only for users of High Availability for Asterisk (HAAst). (I assume that if you had a full backup you would have already tried to

I've been searching the Internet for information regarding the replacement of astdb with a modern sql engine. There are several reasons one would like to do this. First of all, external applications have a hard time reading/writing to the now-old astdb format. Also (and this is what interests me most), the sql astdb could easily be clustered throughout several servers (I'm looking for a

On Tue, Jul 11, 2023 at 3:40 PM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Jul 11, 2023 at 3:38 PM TTT <lists at telium.io> wrote: > >> That answers part two…but is there any mapping of AMI version to Asterisk >> versions? >> > > No, there is not. > I can say that Asterisk 13 is 2.x.x though because I just looked, so you can use the

It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the entire SIP header for a channel. I also read (on stackoverflow) that the PJSIP_HEADER function will only return the headers from the INVITE of the inbound channel. If that’s correct, how would I get the headers from the outbound channel (second leg of the bridged call) INVITE ? Or will PJSIP_HEADERS() in fact return the

On Mon, Jun 26, 2023 at 10:57 AM TTT <lists at telium.io> wrote: > 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

Is there any way to get Asterisk to reload the /var/lib/asterisk/astdb file? It seems to only read it on startup. Thanks.

On Wed, Jun 7, 2023 at 12:04 PM TTT <lists at telium.io> wrote: > Ok that worked. > > > > Since I have not declared a statis app called “test”, does that mean any > non-existent app name on the URL will subscribe to all system events? (Or > is test a built-in app name) > Applications are not declared or configured anywhere. The act of connecting a websocket with an

I tried the command below (with subscribeAll=yes). I made a couple of calls but didn’t see any events. Should I see events? From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua C. Colp Sent: Tuesday, June 6, 2023 6:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users]

On Mon, Jun 26, 2023 at 4:04 PM TTT <lists at 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 also read (on stackoverflow) that the > PJSIP_HEADER function will only return the headers from the INVITE of the > *inbound* channel. > > > > If that’s correct, how would I get the headers from

On Tue, Aug 8, 2023 at 12:44 PM TTT <lists at telium.io> wrote: > 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

Hi listers, We want to implement one call center with asterisk. The idea is it should be scalable, with openser as an dispatcher and bunch of asterisk servers to do ACD, Queues, Agents things... Easy to say :( Look closely to the current asterisk, we do see some problem: - SIP registrations was stored in astdb. - And queue members also was stored in astdb. - ... asterisk was built as

On Mon, Jun 26, 2023 at 4:35 PM TTT <lists at telium.io> wrote: > I think that’s getting me close. I’m trying to get (or recreate) the FROM > and TO lines of the header, from a system running PJSIP. I think if I use > CHANNEL to get local_uri and local_tag I can recreate a FROM line like: > > *FROM=<URI>;tag=TAG* > > > > And if I use CHANNEL to get

How do you load the contact list? It's a database? Sqlite3? https://wiki.asterisk.org/wiki/display/AST/SQLite3+astdb+back-end I'm not clear on what this specific database does. If it's not this specific database which has contact information, which database does? thanks, Thufir

On Sat, Jun 17, 2023 at 2:55 PM TTT <lists at telium.io> wrote: > 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. >

On Tue, Jul 11, 2023 at 3:38 PM TTT <lists at telium.io> wrote: > That answers part two…but is there any mapping of AMI version to Asterisk > versions? > No, there is not. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL:

I’ve been looking through the docs (near https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Applications+REST+API) and am searching for a list of events I can subscribe to. Is this list published? Or can I query the ARI for a list of available events? Thanks Brian From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua C. Colp Sent: Wednesday,

Ok – so if I forgot to add “security” to the read= line in manager.conf for this user, will that cause the user to be unable to subscribe to the “security” events upon login? (in other words, although subscribed at login, no security events will be shown to this user) Thanks From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua C. Colp Sent: Tuesday,

Is it possible to simply send a message to appear as an AMI message/event, from the dialplan? For example exten =>123,1,ami(myEvent, param1, param2) and in the AMI a message appears like: Event: myEvent Privilege: call,all Channel: PJSIP/misspiggy-00000001 Uniqueid: 1368479157.3 ChannelState: 3 ChannelStateDesc: Up CallerIDNum: 657-5309 CallerIDName: Miss Piggy

Hi list. I use sorcery to configure an astdb backend to my pjsip endpoints. This works well, but it would be even better if I could set attribute defaults for these endpoints in the config file. The way I do it now forces me to store all endpoint attributes for each endpoint, even when most of them are effectively defaults. If I need to change one, I need to update each endpoint in the astdb.