Archives : October-2018
FOSDEM is one of the worlds premier meetings of free software developers, with over five thousand people attending each year.FOSDEM 2019 takes place 2-3 February 2019 in Brussels, Belgium.https://fosdem.org This email contains information about: – Real-T..
guys, Iam trying to get current time with milliseconds with STRFTIME ins Asterisk 13:${STRFTIME(${EPOCH},America/Toronto,%F %T.%3q)}But I am getting in all cases:2018-10-29 09:20:46.000Is this because epoch isn’t saving milliseconds or wrong vers..
After originating a PJSIP call, I need to get the channel for that call, so I can end it later in a hangup handler. So I use this:https://wiki.asterisk.org/wiki/display/AST/Function_CHANNELSIn this bit of dialplan:same => n,Originate(PJSIP/0203123456@voipfone-205,exten,bcab-bridge-conference,s,1)s..
Im setting up a new cluster that must replace several old Asterisk instances. For various reasons, this new cluster must use chan_sip (migration to PJSIPis planned in a later phase).This new cluster uses VRRP in active/passive mode:- at any time, o..
i have webrtc client chrome69/jssip which is connecting to asterisk 13.23.1/pjsip i have strange problem where pjsip aor stays in status created sip trace on asterisk looks ok. do you think if this can be bug? test*CLI> pjsip show aors A..
Im testing an Asterisk instance. At the moment, Im focusing on its capability to receive and challenge incoming SIP Registrations.For various reasons, I would prefer to use SIPp instead of Asterisk to act as SIP Client.Has someone successfully done t..
Asterisk 16.0, PJSIPFor the first caller to a conference, I want to dial out and bridge that conference to a new PJSIP external call.For the next callers, I just want them to join the local Asterisk conference.After the last caller leaves the conferen..
I am currently evaluating asterisk 16. I have noticed an issue using application playback. The beginning and the end of the audio file are missing. If I use answer and wait(1) before playback, the beginning is correct. I am using chan_sip, if this..
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 conn..
I wrote some code that connects to the Asterisk AMI to see if the channel is up by doing an Action status followed by a Channel: $CHANNEL. Most of the time if the channel is *NOT* up I will get:line9: Response: Error line10: Message: No such channel..