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I am having a issues with adding a analog card to my dell 2800. I already have a t1 card installed and running fine but when I install the analog card asterisk will not start (ztcfg fails). I have determined it is because of a IRQ problem and have decided to get a new server. Can anyone suggest a server grade setup that supports this? I would rather not buy a machine that will be unstable. I

Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0....

I am trying to tweak my dial plan and I am running into a problem. Sometimes my VoIP out bound calls do not complete on overseas calls(busy or just a hang-up). Is there a way in the dial plan to automatically dial out of my PRI when something like this happens. Either by time limit by a failure event? Any point in the right direction would be great Thanks, CLI output (cleansed to protect the

All, I am setting up asterisk on a nslu2 (Linksys) using unslug. Everything is working great except that I have a polycom 301 and I cannot get the message indicator to work. I have created the users and mailbox in users.conf and I can manually dial the mailbox (*986000). Last thing is I am not using config files for the polycom just web browser. Can anyone point me in the right direction I

I am trying to configure MeetMe so that external callers can enter the conference rooms after an IVR menu. I have created Conf rooms for all internal Ext's with a prefix of 8. When I call into the system from my vonage trunck the IVR picks up but will not let me dial a conf room. It tells me it is a invalid extension. Can anyone help with a sample conf on this? Thanks, RC

Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

Same thing here. I had this problem awhile ago and made this workaround. Going to another trunk does not work because they are answering and not sending a error code. If you are using AAH code then this waits 10 seconds on your Voip then times out and goes to PSTN. You can modify for your needs The pertinent line is 14 in macro-dialout-trunk I am going to clean it up and repost under my

I want to say thanks to everyone for the help so far. I figured out a way to modify some AAH code that worked for me (well sort of). The line I modified is s,14 in macro-dialout-trunk. Then I just added a variable and passed it from 9_outside. I just have one last problem. This waits for an answer not ringing. So if the called party has a long ring to voice mail the call is dropped and goes