Archives : August-2016
If you are happy with the way blacklisting with the Asterisk database works, how about a shell script that loads all of the entries in your blacklist file?Other alternatives involve modifying your dial plan. If you are comfortable with that then ..
Tomorrow, Sunday, community services may have intermittent availability due to maintenance. This maintenance will begin at approximately 2:00 PM CDT[1] and should be complete before 8:00 PMCDT.The affected services include at least:* git.asterisk.o..
to everybody,My IAX is not working, When I type reload IAX it returns me:AsteriskSlave*CLI> iax2 reload== Parsing /etc/asterisk/iax.conf: Found== Parsing /etc/asterisk/users.conf: Found[Aug 26 10:05:04] NOTICE[18078]: chan_iax2.c:13546 set_config:Ignor..
I dial two destination like thisDial(PJSIP/endpoint1/sip:${EXTEN}@${IPA}&PJSIP/endpoint1/sip:${EXTEN}@${IPB})But I need the audio from one of them to be heard by the caller. None gets heard. I switch the order but nothing. How I get the audio for..
I´m trying to get TLS to work with asterisk and client phones, and all I´m getting from asterisk is [Aug 23 11:46:42] WARNING[1170]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason..
Im having an issue with some Snom 300s on a server running Asterisk version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1.There is _*NO NAT*_ involved.Phones and server are plugged into the same network switch, all on the same IP range.The server is runn..
Heres a weirdness – I got a call from someone who couldnt get to my info line earlier, I tried it and it was busy tone.Being on a layby beside a road on a mobile on a long journey, my only real option was a remote server reboot so I couldnt diagn..
If you, like me, have a tendency to leave things until the last minute, you may be in danger of missing out on AstriCon this year – as it is a little earlier than usual, September 27-29 in Glendale, AZ. www.astricon.netThree days of nothing but Asterisk-the..
I am searching a way to dial a SIP peer, and if it does not answer within 20 seconds, play an announcement to the caller. This means that the caller would hear a ring tone for 20 seconds, and only then hear the announcement if the callee did not answe..
Im currently using Asterisk 11.7.0.The issue currently Im facing in Asterisk realtime sip_buddies table i.e. if I try to unregister the extension, ipaddr, port, regseconds, fullcontact, useragent and lastms remain still populated with data unless..