Category : Asterisk Users
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https://github.com/asterisk/asterisk/blob/master/LICENSE#L48 broken(PS I hope I never find a bug to report, because I dont use Github… embrace, extend, extinguish is still alive..
I am banging my head.Stock asterisk on Ubuntu 22.04 (Jammy) installs and works fine, but I want to update the source code.I use this configure line./configure LDFLAGS=-z muldefs –libdir=/usr/lib/x86_64-linux-gnu–with-unixodbc=$(odbc_config –include-prefi..
Asterisk 16.28.0 in use. PJSIP in use Two endpoints Both using IPv6 One Endpoint on UDP, the other via TLS. Both with: t38_udptl=yes ;fax_detect=yes ;fax_detect_timeout=30 rtp_ipv6=yes Both sides are T.38 capable and detect fax tone so no need for ..
What can I do to troubleshoot this error? The machine is fine. I can do isql anydatasource and it works module load res_odbc.so Unable to load module res_odbc.so Command module load res_odbc.so failed. [Apr 23 20:25:59] ERROR[28838]: loader.c:283 module_load_err..
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In order to reduce the amount of system maintenance and administration that needs to be done by the Asterisk team at Sangoma, weve decided to move capabilities such as issue tracking, code management/review and documentation/wiki to hosted solutio..
my logs are flooded with: WARNING: The stasis/m:cdr:aggregator-00000005 task processor queue reached 5000 scheduled tasks again. and then, when call came, I got this: ERROR: Oh dear… we couldnt allocate a port for RTP instance 0x6e1e680fd670 WARNI..
We are moving from an older asterisk/SIP to a newer (18+) asterisk/PJSIP and trying to figure out how to add [identity] header when originating a call from AMI/PAMI. In the older version we would just set a variable like this:$action = new OriginateAction(SIP/….);$action->setVariable(__SIPADDHEADER51,Identi..
I want to configure communication with my phone provider using TLS for all the obvious reasons. Since Im behind a firewall, Ill be needing to do it with NAT. There are examples of UDP plus NAT in pjsip.conf, but none for TLS plus NAT. Would it be corr..
Weve been using Asterisk 16 for a while now, and tried turning on send_rpid = yes in my pjsip config for end points. This solves a problem were having where attended transfers arent updating the CallerID when the transfer is complete (it would s..