Archives : July-2023
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The Asterisk Development Team would like to announce security release Asterisk 19.8.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/19.8.1 and https://downloads.asterisk.org/pub/telephony/aster..
The Asterisk Development Team would like to announce security release Asterisk 18.18.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.18.1 and https://downloads.asterisk.org/pub/telephony/aster..
The Asterisk Development Team would like to announce security release Asterisk 16.30.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/16.30.1 and https://downloads.asterisk.org/pub/telephony/aster..
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..
Ive been having a serious issue the past couple weeks where many users devices show up as Unavailable according to PJSIP. The underlying issue is that res_pjsip thinks there are no available contacts for the device, and in the normal course of operati..
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 ..