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hello list, i have problem when i am useing wrapuptime with agents.conf my agents.conf looks like this [agents] autologoff=15 musiconhold => default wrapuptime=50000 group=1 agent => 1001,4321,Mark Spencer recordagentcalls=yes my aim is every call needs have wrapuptime of 5000 ms but when ever a call comes its directly connecting not wating any more. your views will be highly

Dear freinds, The pl script file which is available in the asterisk monitoring section of the voip-info.com expects username ,password and host parameters . Which one we should provied is the acconts we registered for asterisk or any thing else your help will be immensely appreciated . with regards rk _______________________________________________________ Too much spam in your inbox?

Hello friends, i make a call through queue to the agent when agent lifts the call it gives one side voice and i get this message in the debug chan_sip.c:1880 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) in my sip.conf iam allowing only ulaw can any body suggest where i may be missing with regards RK

Hello list, There seem to be some problem with the ACD of asterisk where when we use this parameter in queues.conf . We could not get any announcement as expected. Iam useing the latest CVS-head Even weight also doesnot seem to work properly I tried like this where we have two queues one with 100 weight and another with 200 as weight when both enter into the queue when queue is empty when

hello list , my set up is like this ip device -->ser ---> asterisk(astcc) --> pstn gatewsy my asterisk version is 1.0.2 iam using the ser as registration and asterisk aa the prepaid one with the help of the astcc. now my problem is the destination people i.e the pstn line s are listening low voice and also the blurr sound quality along with the audio of the ip device at