We have just upgraded a server to Ubuntu 18.04, running kernel4.15.0-142-generic. Compiling DAHDI 3.0.0 fails with the error below, and3.1.0 gives the same error. The gcc compiler is version 7.5.0. Would anyone know what the solution is, or should..
Author : David Cunningham
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We see there is addition of a faxdetect_timeout option and fix for FAXOPT(faxdetect) noted in the Asterisk 13 change log:https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/asterisk-certified-13.8-cert2-summary.htmlWould anyone..
We have aproblem with a SIP doorbell device which sends media one way only, and NAT at the receiving device.When the doorbell button is pressed it makes a call to a configured destination. Since the doorbell only sends and doesnt receive it sends ..
We have an Asterisk server with two public IP addresses, lets say 1.1.1.1and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call dialled from Asterisk to an external destination. The external destination sees the SIP packet as com..
We have an Asterisk 11.3 server where we want log rotation handled purely by Linuxs logrotate, and not by Asterisk. To this end weve configured the[general] action of /etc/asterisk/logger.conf with:rotatestrategy = noneHowever, an asterisk -rx log..
Voisonics is hiring a VoIP support engineer to assist our customers running Asterisk based hosted PBX platforms. This is a part-time contract work-from-home position.For communication reasons were looking for someone in a timezone encompassing Far E..
This has a bit of a long explanation… ultimately the question is why adding a section to sip.conf made a difference to One Touch Record.Were implementing a recording toggle using the Record button on a SIPtelephone and Asterisks One Touch Record feat..
Does anyone know of a way to play different music on hold depending on which party puts the call on hold?We can specify the music on hold per channel, but that doesnt do what is needed. We want to play one music if the caller puts the call on hold, ..
We have a need to record audio and allow the user to press any DTMF key to end the recording. Currently were using the AGI command record filewhich does allow us to specify which DTMF keys can end the recording.However we also need to know *which* ..
Can anyone help with an issue regarding the H264 profile level being passed through Asterisk? We have a video call like this:Caller A -> Asterisk -> Called BCaller As INVITE SDP offers profile-level-idB801f, and Called Breplies a 200 OK containing profile-level-idB8..