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..
all I know, a fairly old asterisk installation. Is there any way to debug T.38 handshaking issues? We have a C3 Voice Switch with link to the Asterisk server. I see this Dialogue: C3 => Asterisk => Invite g711 Invite T.38 Version:0 RateManagement:transferred..
Im building a new Asterisk system from source on Debian Stretch. My building script fails as package libmyodbc is currently missing from Debian Stretch repo.Is there a work around this without leaving MySQL/MariaDB galaxy ?Be..
I just started with setting up a new asterisk system, that will operate on a sip trunk, but I wonder, how to transfer the calls to different extensions, because all calls appear as being send to the base number of the trunk.E.g. given the trunk ra..