Archives : October-2022
We have a problem where Asterisk is resetting the CSeq on a re-INVITE, and the phone receiving the re-INVITE is rejecting it, probably as a result of that. Would anyone be able to offer any insight please?The scenario is:Phone A makes call 1 to Aster..
Bringing 80 devices into a conf bridge with CPU: AMD Phenom(tm) II X6 1045TProcessor at 2.7G and audio is reported as staticy or not the best audio quality.Network is r8169 0000:02:00.0 eth0: RTL8168e/8111Link is 1G.Asterisk 18.14.0I would think t..
The Asterisk Development Team would like to announce the release of Asterisk 20.0.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 20.0.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 19.7.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 19.7.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 18.15.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 18.15.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 16.29.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 16.29.0 resolves several issues repor..
Has there been issues where once in a while RTP audio does not work ?Example: connection to Cisco call manager – works mostly all the time.once in a great while – person does not hear the beep when calling in. once in a great while – person they h..
, Im using Asterisk 11.25.0 and would like to set anonymous@anonymous.invalid as outgoing caller ID via SIP: Set(CALLERID(num)=anonymous@anonymous.invalid) However, when I look at the outgoing packet with tcpdump I see that the @ is not being transmit..
ANyone ever ran into a situation when Call coming from Call Manager into asterisk, is successful coming across – but no Audio ???But then the next call – audio is heard – its once in a great while no audio – most time it works.Anything I might look ..
we are migrating from chan_sip to pjsip
i want logs like this about pjsip endpoints
[Oct 14 17:20:36] NOTICE[35629] chan_sip.c: Peer endpoint22 is now Reachable. (15ms / 2000ms)
is it possible?
thanks
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