search for: vvvvr

...o problems here. I *think* by the FreePBX convention (?) one can not call the 'device' number/reg directly, only the 'user' extension [i actually tried dialing to one of the 'device' SIP reg numbers, 'cannot be completed as dialed' was the answer, and same in the -vvvvr output; the -vvvvr output actually suggests one side RTP is passed, but the other is not, if I read this correctly (on 'normal' calls, both sides RTP is shown 'passed' in the log). The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets);...

...no problems here. I *think* by the FreePBX convention (?) one can not call the 'device' number/reg directly, only the 'user' extension [i actually tried dialing to one of the 'device' SIP reg numbers, 'cannot be completed as dialed' was the answer, and same in the -vvvvr output; the -vvvvr output actually suggests one side RTP is passed, but the other is not, if I read this correctly (on 'normal' calls, both sides RTP is shown 'passed' in the log). The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets);...

Thanks for the speedy response, Danny. So you recommend I run something like: asterisk -vvvvr | tee ast-help.txt Then when I need help on a command I request it on the command line, exit to the shell, edit (or whatever) the .txt file to find the command syntax I am looking for, then re-enter the asterisk cli? Kind of defeats the purpose of 'online help' doesn't it? Not tr...

Hi all. I'm a very computer literate person, but an a bit of a noob when it comes to linux etc. I've a got a brand new PC. I've stuck in a TDM400P, with one FXO port. I've installed Fedora Core 3. Everything good so far. Now for Asterisk. GO to a shell then... md src cd src I'm now at /home/alan/src. Type the appropriate cvs commands (as per asterisk.org), and now i have

Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detect *any* form of incoming call. when I use an outside phone to call the E1 connected to the sangoma a102, I instantly get a fast busy tone. My /etc/zaptel.conf is: loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:1 bus:4 span: 1] span=1,0,0,ccs,hdb3 bchan=1-15,17-31

Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards. There is nothing in the logs and nothing on the console, the call just seems to 'go away'! Can anyone