Category : Asterisk Users
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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 “mana..
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 AMIversion increase with each Asterisk version increase in the future? T..
my asterisk is working fine, I am just confused why, on the server I see private IP address of an endpoint: WARNING: Retransmission timeout reached on transmission 0_252301488@10.1.3.8 for seqno 2 (Critical Response) the IP 10.1.3.8 is a phone beh..
I need to use app_macro, but it seems to be absent from asterisk 16.30.1Is there a w..
I am using Asterisk 16.30 inside Freepbx, with commercial modules, purchased from Sangoma and Symphony. After a few hours my memory usage reaches 900 GB, no kidding, in a box with 1 TB of RAM.The question is: how can I determine what is causing the mem..
Hi.I have run into a problem compiling dahdi-linux in kernel 6.1.0-10.Apparently there was a change, so I found a patch to fix stdbool.h but now I have an implicit declaration of pci_alloc_consistent in drivers/dahdi/wct4xxp/base.c I dont see any ot..
The following AMI command works well for all of the information I want:action: Getvar actionid: act1channel: PJSIP/Twilio-NA-W-3-In-00000028Variable: CHANNEL(pjsip,XXXX)Where XXXX can be one of the many available items.However, when I create a call f..
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 thou..
I have an AGI script written in PHP that worked great with Asterisk 13.Im 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 likeVERB..
I finally got to look at chan_sip to chan_pjsip migration again. This time I’m having problems with influencing codec selection on originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only works on outbound (called) channel and has no aff..