It is with great pleasure I wish to inform you of the first beta release of the new Asterisk 15 branch. Its a very exciting time to be a user of Asterisk! Asterisk 15 is arguably the biggest release of Asterisk that has happened in the last 10 or..
The Asterisk Development Team would like to announce the first beta of Asterisk 15.0.0. This beta is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 15.0.0-beta1 resolves several iss..
Scenario:Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to 192.168.254.1:5060) is behind a NAT, acting as a client to our ITSPs SIP server. But also, this Asterisk is server for various VoIP telephones. Acoording to Asterisks wiki, the transp..
I want to use corosync and installed it via ubuntu repository. I guess there is a version 1 and 2 of corosync. For some reason ./configure for Asterisk (13) isnt recognizing I have corosync installed. I cant enable the res_corosync module in menuselect…
Is very hard to give a suggestion without more information, when call fromCME to Asterisk no voice is detected on both path? how about to collect traffic information between Cisco an Asterisk?Without a call trace andanalyzing with Cisco partner or some..
A few months ago we upgraded our server from Asterisk 220.127.116.11 on CentOS 5.9to Asterisk 13.13.1 on CentOS 7. We are still using SIP not PJSIP.Since the upgrade our remote users conversions are choppy.Here is what my sip.conf looks like for the us..
guys,Could you please let me know whether the latest Asterisk has a support for inbound UPDATE ?In my case, the carrier is sending an UPDATE to change the codec which is replied by 5xx from Asterisk 11.17..
Ok, so a few years ago, when 13 first came out, I was having a core dump (crash) issue with asterisk 13. I worked with Josh some and even used my Digium subscription for support. Never was able to get it fixed at that time so let it go. Well now I..
While trying to use direct_media Im seeing RTP payload mismatches after succesful reinvites.Initial INVITE from endpoint A to asterisk has rfc4733 DMTFm=audio 35648 RTP/AVP 9 8 111 96a=rtpmap:96 telephone-event/8000a=fmtp:96 0-16From asterisk to upstr..
I am using Asterisk 13.9 and using originate with early_media=true.I redirect these calls to an app that I wrote and it just write down audio before the answer. When I detect frame->subclass.integer =AST_CONTROL_ANSWER, app returns to asterisk nor..