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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

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

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:

I have a Vitelity DID which generally works, but calls from a particular caller do not reach it. Vitelity has thus far disavowed any responsibility for working through this problem. I recognize that some action might be required by another provider which is outside Vitelity's control, but it seems that they should at least be trying to help resolve the problem by helping me determine

So, I have been testing ekiga 3.0 with Asterisk, and sadly, it don't work. I am told by the ekiga devs in http://bugzilla.gnome.org/show_bug.cgi?id=553595 and http://bugzilla.gnome.org/show_bug.cgi?id=553810 that the problem is that Asterisk does not support SIP forking. The issue is that I have multiple addresses on my workstation: 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500