I am getting this:make gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC-O2 -MD -MT pritest.o -MF .pritest.o.d -MP -c -o pritest.o pritest.c pritest.c:50:28: fatal error: dahdi/tonezone.h: No such file or directory #include I am us..
Author : Jerry Geis
I am using the AMI interface to start calls.At one point I have a 10 second delay Wait(10) in the dialplan… During this time it seems that if I then connect with the manager interface and place a call that nothing happens till the 10 seconds is done…
I am running 11.20.0 (64 bit) as a user other than root and using the Console/dsp port (soundcard) output and HDMI.I am getting a warble or clicking noise on the audio. Im connected into the pulseaudio for the logged in user. Pulseaudio works fine ..
Do polycom phones not LIKE using something other than port 5060 ???I have five of them behind a firewall and my asterisk server is remote. Other devices are registering just fine, just not my polycom phones.Today I notices that ONE registered, but..
I have 4 Polycom phones behind a firewall. I cannot get them to dial as is says URL Disabled so its not registered to my server.I have tried nat=no, nat=force_rport,comedia still no go.I have sip.conf set for bindaddr= my server IP.I have uncommen..
I am using CentOS 6.7 64 bit and sox to convert gsm files to wav.When I do that I used to get a header like:RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 HzNow I get a header like:RIFF (little-endian) data, WAVE audio, GSM 6…
I have a setup where I have polycom phones in an office, behind firewall, going out to a server located elsewhere. I have set nat=force_rport,comedia for my phones.so if I call OUT to my cell I get audio both ways and the call is fine.My issue is..
I have a polycom phone behind a firewall. The phone registers – but I only hear half channel audio.I have tried nat=yes, nat=force_rport,comedia and nat=autio_force_rport,auto_comedia (reloading asterisk every time).made no difference.How might I ..
I am looking for a push to talk solution does anyone know of a good PTT phone one that works with asterisk.Im not talking about polycom fake PPT… Im talking about a real call into Asterisk and having to push a button on a headset or the phone to actua..
I would like to figure out using confbridge how to play a file after the conf is built.not really a per user thing – just conf is up and ready and need to play a file to all in the conference.(I am creating my conf on the fly and bringing in other devi..