Dialout redundancy using this method works perfect. I've been using this method for some time now. I currently have two IAX2 providers and plan to get another backup as well (In addition to me getting my Digium cards tomorrow that'll be another backup.)
That's great for outgoing calls, but... I'm trying to figure out the best approach to use for incoming calls. I currently have a VP phone number, it's the only incoming number I have for the other voip providers I have don't offer local termination (or any at all for that matter). We have a POTS line from Verizon and we'd like to continue using that phone number. Originally we were just going to forward that phone number to VP. But what happens if VP goes down? I figure in that case (and we'd have to get in touch with VP if they will forward to another number if they're done), to then forward to another voip / pots line that we have. Is there any other approach we can use to do this? Possibly, a service that'll offer something like: Transfer to 1609xxxxxxx but if busy, forward to 1609xxxxxxx, etc. and so on? In addition does anyone know where I might be able to port my number to that supports transferring instead of forwarding? I currently have Verizon and they said we need a CustoFlex plan which will only support 6 "forwards" so if 7 callers call in, the 7th will get a busy signal. -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Franks Sent: Sunday, February 08, 2004 3:15 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] dialout redunancy. You will need to set priorities for each one. For example: exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91NXXNXXXXXX,2,Playback(pstnallbusy) exten => _91NXXNXXXXXX,3,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}} exten => _91NXXNXXXXXX,4,Congestion Basically what happens here, is I try to put it out on the Verizon POTS lines first, then if that doesn't work, I play a message saying all the lines are busy, hold if the call is important (it's now billable), the user holds, and it goes to voicepulse. You could get rid of the All Busy message if you wanted, I just like to know that the call is going to be billed (since I have unlimited LD on my POTS lines). If that fails, It plays a fast busy. You can also do a qualify in your iax.conf and sip entries to know whether they are reachable before trying the call. Read up on qualify to find out how to do it for your needs. Brent -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Sent: Sunday, February 08, 2004 2:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] dialout redunancy. Hi, How do I get asterisk to use an alternate outbound provider in the event my primary IAX provider goes down. I currently have an IAX provider that is having issues, so I signed up with a sip provider for a backup. I added the sip provider info into the extensions.conf file as the second outbound entry, but asterisk still tries to call the iax provider 1st and since the call is incomplete the end-user hangs up. Any ideas would be helpful. Thanks John Bittner Simlab.net _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
