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

On Tue, Jul 11, 2023 at 3:40 PM Joshua C. Colp wrote: > On Tue, Jul 11, 2023 at 3:38 PM TTT 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

Re: [asterisk-users] AMI versions

On Tue, Jul 11, 2023 at 3:38 PM TTT 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 --

Re: [asterisk-users] AMI versions

That answers part two…but is there any mapping of AMI version to Asterisk versions? From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Bright Sent: Tuesday, July 11, 2023 11:58 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AMI

Re: [asterisk-users] AMI 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 <[li...@telium.io](mailto:On Tue, Jul 11, 2023 at 11:54 AM, TTT < wrote: > Is there a web page that lists the AMI versions

[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] AMI vs. Dialplan Originate

You could do the old school method and create and move a .call file from your dialplan. exten => writefile,1,NoOP() same => n,Set(CALLFILE=/var/spool/asterisk/tmp/${FileName}-${ARG1}.call) same => n,Set(FILE(${CALLFILE},,,al,u)=Channel: SIP/bob) same => n,Set(FILE(${CALLFILE},,,al,u)=WaitTime:

Re: [asterisk-users] AMI vs. Dialplan Originate

On Tuesday 22 September 2020 at 13:27:27, Joshua C. Colp wrote: > On Tue, Sep 22, 2020 at 7:37 AM Antony Stone wrote: > > Hi. > > > > (Asterisk 16.2.1) > > > > I'm using AMI Originate to initiate calls, and I'm passing some > > additional data in to the dialplan context using the Variable: > >

Re: [asterisk-users] AMI vs. Dialplan Originate

On Tue, Sep 22, 2020 at 7:37 AM Antony Stone < antony.st...@asterisk.open.source.it> wrote: > Hi. > > (Asterisk 16.2.1) > > I'm using AMI Originate to initiate calls, and I'm passing some additional > data in to the dialplan context using the Variable: parameter. Works fine. > > >

[asterisk-users] AMI vs. Dialplan Originate

Hi. (Asterisk 16.2.1) I'm using AMI Originate to initiate calls, and I'm passing some additional data in to the dialplan context using the Variable: parameter. Works fine. https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+ManagerAction_Originate Now I need to do the same thing but from

Re: [asterisk-users] AMI not responding correctly

> -Original Message- > From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On > Behalf Of Antony Stone Sent: Wednesday, May 29, 2019 4:17 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] AMI not > responding co

Re: [asterisk-users] AMI not responding correctly

-Original Message- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Antony Stone Sent: Wednesday, May 29, 2019 4:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI not responding correctly On Wednesday 29 May 2019

Re: [asterisk-users] AMI not responding correctly

On Wednesday 29 May 2019 at 22:01:11, Jason wrote: > I am communicating How? > with Asterisk 13.18.3 over the AMI and issue the command: > > ActionID: 11 > Action: command > Command: core show calls > > And the response I get is: > > Response: Follows > Privilege: Command > ActionID: 11 >

[asterisk-users] AMI not responding correctly

I am communicating with Asterisk 13.18.3 over the AMI and issue the command: ActionID: 11 Action: command Command: core show calls And the response I get is: Response: Follows Privilege: Command ActionID: 11 --END COMMAND- But where is the call data? What is going wrong on

Re: [asterisk-users] AMI mulitple calls quickly

Thanks all for the suggestions. 1) ActionID. I was not using, so that could definitely have helped. 2) I was using Async already. 3) I changed to open multiple AMI connections - Worked like a champ. 4) Used to use call files but slow to first action... So changed to AMI long time ago. Thanks,

Re: [asterisk-users] AMI mulitple calls quickly

On 3/12/19 11:03 AM, Steve Edwards wrote: On Mon, 11 Mar 2019, Jerry Geis wrote: If I use the AMI interface to originate a call, close the connection, open another connection etc...This works. but is slow... Would opening multiple AMI connections be an option?     You should be able to

Re: [asterisk-users] AMI mulitple calls quickly

On Mon, 11 Mar 2019, Jerry Geis wrote: If I use the AMI interface to originate a call, close the connection, open another connection etc...This works. but is slow... Would opening multiple AMI connections be an option? -- Thanks in advance,

Re: [asterisk-users] AMI mulitple calls quickly

On Tuesday 12 March 2019 at 01:19:37, Jerry Geis wrote: > Lets say I have to make 40 phone calls quickly. > > If I use the AMI interface to originate a call, close the connection, open > another connection etc... > This works. but is slow... How about using call files instead?

[asterisk-users] AMI mulitple calls quickly

Lets say I have to make 40 phone calls quickly. If I use the AMI interface to originate a call, close the connection, open another connection etc... This works. but is slow... If I open the AMI interface and originate a call - DONT close the interface , get the response, originate another call,

Re: [asterisk-users] AMI not listening on secondary IP address?

On Tuesday 23 October 2018 at 12:51:56, Doug Lytle wrote: > >>> No, it's not a firewall problem; I've currently allowed connections to > >>> 5038 > > Antony, > > Do you have any deny/permit section in the manager.conf that would need to > be adjusted? No, and since I posted this, I've found

Re: [asterisk-users] AMI not listening on secondary IP address?

>>> No, it's not a firewall problem; I've currently allowed connections to 5038 Antony, Do you have any deny/permit section in the manager.conf that would need to be adjusted? Doug -- _ -- Bandwidth and Colocation Provided

[asterisk-users] AMI not listening on secondary IP address?

Hi. I have three servers running corosync and pacemaker, to maintain a floating address between them. This is working fine, and I can, for example, SSH to the floating address and get to whichever server has the address at the time. I am trying to connect to the same server (using the same

Re: [asterisk-users] AMI status events with res_fax_spandsp.so

>>> Is anyone else using the AMI with res_fax_spandsp.so for real-time status? >> >> As far as I can tell there is no way to get worthwhile status/progress >> information from AMI when using spandsp. >> >> Neil Youngman > >Thanks, I figured that was the case but wanted to be sure I hadn’t

Re: [asterisk-users] AMI manager logins - omitting from logging output?

On Thursday 07 June 2018 at 10:44:15, Tony Mountifield wrote: > In article <201806070119.51560>, Antony Stone wrote: > > > > Is there any way to tell AMI that I don't want it to log login attempts - > > or, to put it another way, is there any way to tell the logger module to > > ignore AMI? > >

Re: [asterisk-users] AMI manager logins - omitting from logging output?

In article <201806070119.51560.antony.st...@asterisk.open.source.it>, Antony Stone wrote: > Hi. > > Is there any way to eliminate AMI manager logins from the logging output > (without just turning the log level down and thereby losing lots of other > stuff > as well)? > > I'm running

[asterisk-users] AMI manager logins - omitting from logging output?

Hi. Is there any way to eliminate AMI manager logins from the logging output (without just turning the log level down and thereby losing lots of other stuff as well)? I'm running Asterisk 13.14.1 as a backend service to LVS/IPVS, and using the AMI login as the "service alive" check to see

Re: [asterisk-users] AMI status events with res_fax_spandsp.so

>> Is anyone else using the AMI with res_fax_spandsp.so for real-time status? > > As far as I can tell there is no way to get worthwhile status/progress > information from AMI when using spandsp. > >> I am working on migrating a FAX application from res_fax_digium.so to >> res_fax_spandsp.so.

[asterisk-users] AMI status events with res_fax_spandsp.so

Is anyone else using the AMI with res_fax_spandsp.so for real-time status? I am working on migrating a FAX application from res_fax_digium.so to res_fax_spandsp.so. I have noticed that the spandsp module generates far fewer AMI status events than the Digium module and the generated events

Re: [asterisk-users] AMI potential memory leak which may be causing a crash

Unfortunately, upgraded to Asterisk 13.20.0 and we are still seeing strange results in the AMI AsyncAGIExec Result string. First one for the call is successful. Later during the same call, it has characters that would be from some ExternalIVR work for this call. The command initiated was an

Re: [asterisk-users] AMI potential memory leak

Of Dan Cropp Sent: Thursday, March 22, 2018 10:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI potential memory leak We just received a separate call with a Result that seems random... This is on a separate box running Asterisk 14.7.5

Re: [asterisk-users] AMI potential memory leak

- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Thursday, March 22, 2018 9:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI potential memory leak Not sure if this may

Re: [asterisk-users] AMI potential memory leak

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Thursday, March 22, 2018 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI potential memory leak HI Matt, I am trying to replicate this particular problem. We

Re: [asterisk-users] AMI potential memory leak

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Fredrickson Sent: Wednesday, March 21, 2018 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI potential memory leak

Re: [asterisk-users] AMI potential memory leak

On Wed, Mar 21, 2018 at 4:03 PM, Dan Cropp wrote: > We are communicating with Asterisk via AMI. Running Asterisk version > 13.18.5 on an Ubuntu box. > > > > If you look at the event response, the Result field is filled with random > characters. I’m not sure what to do because

[asterisk-users] AMI potential memory leak

We are communicating with Asterisk via AMI. Running Asterisk version 13.18.5 on an Ubuntu box. If you look at the event response, the Result field is filled with random characters. I'm not sure what to do because that is a completely random result. It makes no sense. We send the following

Re: [asterisk-users] AMI column widths

You are using AMI to run CLI commands and that's the problem. Try to use the equivalent AMI actions to get the information that you want. My suggestion : get all channels in use (CoreShowChannels) and then filter just the SIP, since there is not an action to do exactly what you need. On 8 Jul

Re: [asterisk-users] AMI column widths

On Saturday 08 July 2017 at 10:16:19, Antony Stone wrote: > On Saturday 08 July 2017 at 07:15:08, Marcelo Terres wrote: > > There are no sip show channels on AMI. Also, the output that you sent is > > not a AMI output. Are u using AMI ou running commands on console? > > I'm using AMI. > > I

Re: [asterisk-users] AMI column widths

On Saturday 08 July 2017 at 07:15:08, Marcelo Terres wrote: > There are no sip show channels on AMI. Also, the output that you sent is > not a AMI output. Are u using AMI ou running commands on console? I'm using AMI. I have a connection to the Asterisk server on port 5038, initated with:

Re: [asterisk-users] AMI column widths

There are no sip show channels on AMI. Also, the output that you sent is not a AMI output. Are u using AMI ou running commands on console? Running commands on console and parsing the output is the worst way to obtain data, first because it is not easily parseable. Second, it doesn't show you all

Re: [asterisk-users] AMI column widths

On Fri, 7 Jul 2017, Antony Stone wrote: I'm trying to get a list of the channels currently in use on an Asterisk server (1.8.32.1 if it matters) over AMI. Would the AMI 'CoreShowChannel' or the CLI 'core show channels concise' commands help? -- Thanks in advance,

[asterisk-users] AMI column widths

Hi. I'm trying to get a list of the channels currently in use on an Asterisk server (1.8.32.1 if it matters) over AMI. I send the command "sip show channels", and I get back a response along the lines of (* used to protect the innocent...): Peer User/ANR Call ID

[asterisk-users] AMI "Queues" Action

Currently the AMI "Queues" action outputs the same text that the CLI outputs when running a "queue show" command, which does not conform with the AMI spec. It should follow the same format as other AMI actions, structured in a key value list. The "QueueStatus" action outputs the information that

Re: [asterisk-users] AMI Originate not working

Thomas, this code block should work for your Originate case. This code block will dial a local channel where actual leg 1 number is dialed. On Answer of leg1, the leg2 is called. - require_once('phpagi-2.20/phpagi-asmanager.php'); $asm =

[asterisk-users] AMI Originate not working

Hello, I want to call an phone and if phone picked up I want to ring another phone. Or I want to connect to an running channel and then call another phone or move to an ConfBridge Iam using PHP $channel = 'IAX2/556-1696'; or $channel = 'SIP/0019736363636@outbound.patton'; $exten = '';

[asterisk-users] AMI ManagerAction UpdateConfig on file with #include may corrupt the file

I have seen the following scenario that may lead to a corrupted and possibly invalid configuration file after using UpdateConfig through AMI, at least with Asterisk 11.25: There is a configuration file a.conf that contains several sections and also contains a #include "b.conf", which defines a

Re: [asterisk-users] AMI version of CONNECTEDLINE

Hi Jacek, Thank you very much for the suggestion. Using SetVar and CONNECTEDLINE(number) works. On 12 December 2016 at 18:31, Jacek Konieczny wrote: > On 2016-12-12 02:21, David Cunningham wrote: > >> Is there any equivalent of the CONNECTEDLINE function which can be >>

Re: [asterisk-users] AMI version of CONNECTEDLINE

On 2016-12-12 02:21, David Cunningham wrote: Is there any equivalent of the CONNECTEDLINE function which can be called from an application using the AMI? You can use dialplan functions from AMI using GetVar, so this should work: Action: GetVar Variable: CONNECTEDLINE(num) Jacek --

[asterisk-users] AMI version of CONNECTEDLINE

Hello, Is there any equivalent of the CONNECTEDLINE function which can be called from an application using the AMI? Thanks for any ideas. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 --

[asterisk-users] AMI version in Asterisk 14

I noticed that Asterisk 14 has changed the output format for some commands (eg: "Output: "). However, the AMI reports version 2.8.0 which is the same as Asterisk 13 Is that considered a bug? Since the AMI output format has changed, shouldn't the AMI version be incremented? (Makes is hard

[asterisk-users] AMI Originate and 183 response

I to Originate channels using AMI. When the other end indicates the channel is ringing, I need to do some system notification work. Everything works great when the ITSP sends a 180 Ringing response. Through AMI events I see the channel state changed and can do the necessary work. However,

[asterisk-users] AMI issue with Filter

Hello all, I am trying to use the Filter action in AMI to make AMI less chatty by blacklisting some events; and I must be doing something wrong, because if I send something like: Action: Filter ActionID: AID563116752-152218 Operation: Add Filter: !Event: VarSet* Filter: !Event: ExtensionStatus*

Re: [asterisk-users] AMI: check if the user has a Mailbox

On Thu, Apr 21, 2016 at 09:34:47PM +0200, Luca Bertoncello wrote: > On an Asterisk-Server I have some users. Just two of them have a Mailbox. > I want to write a little Web interface to manage many things and I'd like to > have a menu point for the voicemail, but just if the user has a Mailbox.

Re: [asterisk-users] AMI: check if the user has a Mailbox

Hi Luca Would greping for the existence of the mailbox number in /etc/voicemail.conf do the trick? Pete > On 22/04/2016, at 7:34 am, Luca Bertoncello wrote: > > Hi list! > > On an Asterisk-Server I have some users. Just two of them have a Mailbox. > I want to write a

Re: [asterisk-users] AMI: check if the user has a Mailbox

Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support Sent: Thursday, April 21, 2016 4:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] AMI: check if the user has

Re: [asterisk-users] AMI: check if the user has a Mailbox

To: Asterisk Users Subject: [asterisk-users] AMI: check if the user has a Mailbox Hi list! On an Asterisk-Server I have some users. Just two of them have a Mailbox. I want to write a little Web interface to manage many things and I'd like to have a menu point for the voicemail, but just if the user

[asterisk-users] AMI: check if the user has a Mailbox

Hi list! On an Asterisk-Server I have some users. Just two of them have a Mailbox. I want to write a little Web interface to manage many things and I'd like to have a menu point for the voicemail, but just if the user has a Mailbox. I found the AMI-Command MailboxStatus, but it does not return

[asterisk-users] AMI StarAstAPI.php not working for commands like SIPpeers with more output

Hi, Iam using StarAstAPI.php. If Iam sending Action commands like 'reload' everythink works fine. If I send an Action like 'SIPpeers' I get: ["Response:"]=> string(8) " Success" ["ActionID:"]=> string(2) " 2" ["EventList:"]=> string(6) " start" ["Message:"]=> string(29) " Peer

[asterisk-users] AMI event PeerStatus: Address option with res_pjsip

Hello. Continue to move from chan_sip to res_pjsip. For the work of my algorithms is very important to know the IP address of all trunks and endpoints (phones). In the case of chan_sip, I used PeerStatus AMI event through which was received the fact of online/offline and IP Address of

[asterisk-users] AMI Redirect from in a WaitExten

Hi, I have and AMI application that tries to redirect a channel if a certain condition exists. It seemed to work when using Asterisk version 11.14, but now I am trying it with 11.19 and it is not. Here is the scenario: 1. A channel connects to the dialplan and is put into a

[asterisk-users] AMI 'meetme list concise' hanging

I have a problem with AMI 'meetme list concise' hanging. I'm running Asterisk 11.15.1, and PHPAGI 2.20. The AMI stuff is in the file phpagi-asmanager.php, which says it is v 1.10 2005/05/25. Here's the relevant snippet of my PHP code: // get list of conferences if

[asterisk-users] AMI Originate not execution failed extension on Asterisk 13.2

We relay on 'failed' extensions after AMI ORIGINATE command. When moving from Asterisk 1.8.22 to Asterisk 13.2, it has stopped to work. I belive that it is due to a change in pbx.c = ast_pbx_outgoing_exten. Thanks, Valter --

[asterisk-users] AMI Redirect both calls from a bridge

Doe anybody know of a way to redirect both channels from a bridge to different dial plan extensions from the using the AMI. Currently, as soon as I redirect one of the channels the other appears to be dropped and gets reorder tone (congestion, fast busy). I guess what I really need is a way

Re: [asterisk-users] AMI Redirect both calls from a bridge

On 2014-12-17 9:08 AM, Neil Cherry wrote: Doe anybody know of a way to redirect both channels from a bridge to different dial plan extensions from the using the AMI. Currently, as soon as I redirect one of the channels the other appears to be dropped and gets reorder tone (congestion, fast

Re: [asterisk-users] AMI Redirect both calls from a bridge

Hi Neil, Am Mittwoch, den 17.12.2014, 09:08 -0500 schrieb Neil Cherry: Doe anybody know of a way to redirect both channels from a bridge to different dial plan extensions from the using the AMI. Currently, as soon as I redirect one of the channels the other appears to be dropped and gets

Re: [asterisk-users] AMI Redirect both calls from a bridge

On 2014-12-17 9:34 AM, Karsten Wemheuer wrote: Hi Neil, Am Mittwoch, den 17.12.2014, 09:08 -0500 schrieb Neil Cherry: Doe anybody know of a way to redirect both channels from a bridge to different dial plan extensions from the using the AMI. Currently, as soon as I redirect one of the

Re: [asterisk-users] AMI and CDR(answer)

On Thu, Oct 16, 2014 at 4:12 PM, Murthy Gandikota mgandik...@nts.net wrote: in cdr.c void ast_cdr_reset(struct ast_cdr *cdr, struct ast_flags *_flags) { struct ast_cdr *duplicate; struct ast_flags flags = { 0 }; if (_flags)

Re: [asterisk-users] AMI and CDR(answer)

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Friday, October 17, 2014 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI and CDR

Re: [asterisk-users] AMI and CDR(answer)

On Wed, Oct 15, 2014 at 9:31 PM, Murthy Gandikota mgandik...@nts.net wrote: Thanks, Matthew. I think CDR(answer) is, in the end, not very useful to me if it changes from context to context. Suppose from AMI we generate a status I'm not sure what you mean by changes from context to context.

Re: [asterisk-users] AMI and CDR(answer)

-Commercial Discussion Subject: Re: [asterisk-users] AMI and CDR(answer) On Wed, Oct 15, 2014 at 9:31 PM, Murthy Gandikota mgandik...@nts.net wrote: Thanks, Matthew. I think CDR(answer) is, in the end, not very useful to me if it changes from context to context. Suppose from AMI we generate

Re: [asterisk-users] AMI and CDR(answer)

Gandikota Sent: Thursday, October 16, 2014 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI and CDR(answer) Suppose we have a dialplan as follows and sip.conf is set to forward the call to [start] context, the CDR(diposition) in [start] context

Re: [asterisk-users] AMI and CDR(answer)

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy Gandikota Sent: Thursday, October 16, 2014 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI and CDR(answer) Apparently we are calling ResetCDR (not ForkCDR) in the Asterisk 11.5.1

Re: [asterisk-users] AMI and CDR(answer)

-boun...@lists.digium.com] On Behalf Of Murthy Gandikota Sent: Thursday, October 16, 2014 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI and CDR(answer) Hi Matthew, Now that you helped me figure out the root cause of my problem, I am

[asterisk-users] AMI and CDR(answer)

Hi All I am unable to obtain CDR(answer) in AMI. Tried the following: $ telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Asterisk Call Manager/1.0 Action: Login ActionID: 1 Username: admin Secret: secret5 Action: Getvar

Re: [asterisk-users] AMI and CDR(answer)

On Wed, Oct 15, 2014 at 1:44 PM, Murthy Gandikota mgandik...@nts.net wrote: Hi All I am unable to obtain CDR(answer) in AMI. Tried the following: $ telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Asterisk Call Manager/1.0 Action:

Re: [asterisk-users] AMI and CDR(answer)

-Commercial Discussion Subject: Re: [asterisk-users] AMI and CDR(answer) On Wed, Oct 15, 2014 at 1:44 PM, Murthy Gandikota mgandik...@nts.net wrote: Hi All I am unable to obtain CDR(answer) in AMI. Tried the following: $ telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to localhost

Re: [asterisk-users] AMI and CDR(answer)

:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI and CDR(answer) Hi Mathew The channel was answered. I have a small AGI script that would call the getFullVariable(${CDR(answer)}) method in Java and print a Date/Time string. When the AMI connection

Re: [asterisk-users] AMI and CDR(answer)

...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy Gandikota Sent: Wednesday, October 15, 2014 2:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI and CDR(answer) I traced CDR(disposition) which was set to NO ANSWER

Re: [asterisk-users] AMI and CDR(answer)

On Wed, Oct 15, 2014 at 5:10 PM, Murthy Gandikota mgandik...@nts.net wrote: The CDR(disposition) is changing from context to context. Looks like AGI and AMI are in agreement. Still, it is a mystery why the helpful Asterisk folks haven't given us a built-in variable for when the call was first

Re: [asterisk-users] AMI and CDR(answer)

: Wednesday, October 15, 2014 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI and CDR(answer) On Wed, Oct 15, 2014 at 5:10 PM, Murthy Gandikota mgandik...@nts.net wrote: The CDR(disposition) is changing from context to context. Looks like AGI

Re: [asterisk-users] AMI Elastix

Thanks! It Completely resolves troubles 2014-08-18 19:52 GMT+06:00 Mikael Fredin mik...@wiraya.com: On 18 August 2014 15:35, Усин Айбек prince...@gmail.com wrote: Thanks, im find it/ but in win7 it doesnt work. and it work under linux That's because windows terminates newlines

[asterisk-users] AMI CoreShowChannel missing Application field

Asterisk 12.5 The CoreShowChannel event (in response to the CoreShowChannels action) no longer returns the Application field as it did in Asterisk 11. Is this a bug or a feature? -- Mitch -- _ -- Bandwidth and Colocation

Re: [asterisk-users] AMI CoreShowChannel missing Application field

On Fri, Aug 22, 2014 at 1:40 PM, Mitch Claborn mitch...@claborn.net wrote: Asterisk 12.5 The CoreShowChannel event (in response to the CoreShowChannels action) no longer returns the Application field as it did in Asterisk 11. Is this a bug or a feature? Yup, that's a bug. When things got

Re: [asterisk-users] AMI CoreShowChannel missing Application field

On 08/22/2014 02:47 PM, Matthew Jordan wrote: Yup, that's a bug. When things got ported over to hit the cached snapshots of the channels (as opposed to locking the live channel), that field got missed. Please file a bug on issues.asterisk.org. Thanks! Matt

[asterisk-users] AMI Elastix

Hi all! I have trouble with connection to AMI 1.1 wich enabled on Elastix *Asterisk Call Manager/1.1* *Action: Login Username: admin Secret: qweasd123* *Response: Error* *Message: Missing action in request* Elastix versions: * Kernel* * Linux(x86_64)-2.6.18-348.1.1.el5* * Elastix* *

Re: [asterisk-users] AMI Elastix

update new state: * == Client from 192.168.0.95, failed to authenticate in 30 seconds* * == Connect attempt from '192.168.0.95' unable to authenticate* * == Client from 192.168.0.95, failed to authenticate in 30 seconds* * == Connect attempt from '192.168.0.95' unable to authenticate* *--

Re: [asterisk-users] AMI Elastix

On 18 Aug 2014, at 09:27, Усин Айбек prince...@gmail.com wrote: I have trouble with connection to AMI 1.1 wich enabled on Elastix Asterisk Call Manager/1.1 Action: Login Username: admin Secret: qweasd123 Response: Error Message: Missing action in request You are missing the newline

Re: [asterisk-users] AMI Elastix

Thanks, im find it/ but in win7 it doesnt work. and it work under linux 2014-08-18 18:02 GMT+06:00 Steven Howes steve-li...@geekinter.net: On 18 Aug 2014, at 09:27, Усин Айбек prince...@gmail.com wrote: I have trouble with connection to AMI 1.1 wich enabled on Elastix *Asterisk Call

Re: [asterisk-users] AMI Elastix

On 18 August 2014 15:35, Усин Айбек prince...@gmail.com wrote: Thanks, im find it/ but in win7 it doesnt work. and it work under linux That's because windows terminates newlines differently, you should convert newlines under windows to windows format (\r\n). --

[asterisk-users] AMI Originate CDRs

Hello, There seems to be a problem with asterisk cdrs when calls are generated via AMI Originate using Local channels. Asterisk writes CDR as soon as A party off-hooks. Resulting in very inaccurate billsec and duration values. Expected CDR in case of local channel origination should be 2

Re: [asterisk-users] AMI Originate CDRs

Instead of using CDR for this, could you get the info you need using channel event logging (Asterisk CEL)? I have never used it myself - just something I've run across in the past that seems like it might work for this case: https://wiki.asterisk.org/wiki/display/AST/CEL+Design+Goals On Thu,

Re: [asterisk-users] AMI and pyst

On Thu, Apr 10, 2014 at 9:14 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Does anyone on this list use pyst for AMI purposes? If so, can you point me in the direction of some simple examples. There seems to be none anywhere online. Probably doesn't help that I'm not that experienced at

Re: [asterisk-users] AMI and pyst

On 14 April 2014 16:34, Matthew Jordan mjor...@digium.com wrote: On Thu, Apr 10, 2014 at 9:14 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Does anyone on this list use pyst for AMI purposes? If so, can you point me in the direction of some simple examples. There seems to be none anywhere

[asterisk-users] AMI and pyst

Does anyone on this list use pyst for AMI purposes? If so, can you point me in the direction of some simple examples. There seems to be none anywhere online. Probably doesn't help that I'm not that experienced at python but not insurmountably so. Thanks in Advance Ish -- Ishfaq Malik

[asterisk-users] AMI Proxy

Hi people Just having a quick check to see if anyone is using any AMI proxies and which are the most popular. For our purposes it must be able to connect to multiple asterisk instances. Thanks for the help. Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845

Re: [asterisk-users] AMI Proxy

On Mon, Mar 24, 2014 at 6:17 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi people Just having a quick check to see if anyone is using any AMI proxies and which are the most popular. For our purposes it must be able to connect to multiple asterisk instances. All depends on the language you

Re: [asterisk-users] AMI Proxy

On Mon, Mar 24, 2014 at 6:17 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Just having a quick check to see if anyone is using any AMI proxies and On Mon, 24 Mar 2014, Paul Belanger wrote: All depends on the language you want to use. We used starpy for a while, but ended up rewriting our own

[asterisk-users] AMI can not receive events unless keyboard input

Hi, This is evenily. My one Asterisk server and event listen App was working well for several month. Yesterday My event listen app stop work suddenly. I telnet to AMI via localhost 5038 to check the events, and I find Asterisk does not push the events until I press the Enter key. When I press

Re: [asterisk-users] AMI eventmask question

On Thu, Jan 23, 2014 at 9:05 PM, Daniel Jenkins dan.jenkin...@gmail.comwrote: On Thu, Jan 23, 2014 at 8:46 PM, Matthew Jordan mjor...@digium.comwrote: On Thu, Jan 23, 2014 at 9:31 AM, Michelle Dupuis mdup...@ocg.ca wrote: Thanks - I've been through that doc before and couldn't find the

Re: [asterisk-users] AMI eventmask question

On Thu, Jan 23, 2014 at 3:25 AM, Michelle Dupuis mdup...@ocg.ca wrote: Hi I'm creating an AMI client and I only want to get newchannel events (as well as responses to any actions I initiate). What would I set the eventmask to to only get the newchannel events? Are you talking about the

Re: [asterisk-users] AMI eventmask question

[ dan.jenkin...@gmail.com] *Sent:* Thursday, January 23, 2014 9:03 AM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] AMI eventmask question On Thu, Jan 23, 2014 at 3:25 AM, Michelle Dupuis mdup...@ocg.ca wrote: Hi I'm creating an AMI client and I only want to get newchannel

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