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Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me

Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, does anyone know of a good calling card solution for asterisk that is able to do lcr? Does astcc do this? I've been searching around and I can find some lcr modules/apps but none that incorporate prepaid card functionality. Regards, Igor H.

Hello, We've had this problem happen twice with retail customers already and still have no solution. Basically there are times when customers can't get any ring at all. It happens that they call our switch and even though we are receiving ring from the carrier they hear no ring. We have even put a fake-ring(with Rr) back at their request and they are unable to get this ring either. The

Hey guys, I've been having a very odd problem that happens intermittently. I've had this happen with only a couple of providers and somewhat rarely but its to the point now that we need to fix it to be able to do business. The scenario is as follows: We have a DID provider that routes calls to our asterisk boxes and we have an outbound provider to whom we send the calls of the person

Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf