I am trying to get audio to work on AWS using asterisk 18.14.0I have enabled the firewall to allow ALL UDP on AWSMy SIP extension has nat=force_rport,comedia qualify=yes allow=ulaw allow=alaw allow=gsm canreinvite=yesI enable rtp set debug on and ..
I am just doing a basic call in.exten => 140,1,Answer exten => 140,n,Playback(beep)exten => 140,n,Dial(MulticastRTP/basic/239.168.4.90:3040//t(15))exten => 140,n,Hangupthis works – but sometimes I get reports that nothing was heard. Is there anyth..
Hi. I have a dialplan which calls the VoiceMail() application, and Im getting the following behaviour: – if the inbound caller leaves a message, then presses #, and then presses 1 to accept the recording, everything works as expected and the dialp..
I am getting a compile error:gcc -g-Wall -Werror -Wstrict-prototypes -Wmissing-prototypes-Werror=zero-length-bounds -fPIC-O2-MD -MT q921.o -MF .q921.o.d -MP -c-o q921.o q921.c q921.c: In function ‘q921_dump’:q921.c:1333:85: error: array subscr..
AllHow do I restart logging in /var/log/asterisk/messages ?asterisks is still running – but logging stopped. I think a process trimsthe file. How (with stopping and starting) do I get logging to happen again.I see downloads.Asterisk.org has a dahdi rele..
Community,Does anyone know if there is a way to set the http curl timeout value for the res_http_media_cache module? As far as we read the code it is not meant to be changed. We use it to download audio from http servers by Playback ARI Command. It..
Asking because I see there is a new DeadlockStart event added to 18.15.0 but the AMI_VERSION value is stil..
I am trying to use TONE_DETECT in a call that is made into a call centre and placed on hold, when the call has been answered by the agent (denoted by a dial tone of 2 seconds of dual tone 350Hz + 440Hz followed by 4seconds of silence (North Ameri..
Hi. I have a dialplan which accepts an inbound call and dials out to another number, automatically bridging the channels together when the second call is answered. I then have a facility for the caller to put the call on hold (which uses ChannelRedirec..
whats best method from perfomance view to parse this headerP-Asserted-Identity:P-Asserted-Identity:i need+44111222333 (if Privacy: id)PJSIP_PARSE_URI ?STRREPLACE/CUT?FastAGI?other options?th..