Bruce B
2012-Dec-04 20:47 UTC
[asterisk-users] How to check channel status and move on silently?
Hello, I have 10 different routes with few different providers. When I place an international call, I would like the system to try all those routes and place the call through whichever possible. If there is any message but an ANSWER the system should move on to next route. I know this is not the best strategy but there are so many bad routes now-a-days that it's becoming a headache. The only requirement here is to no pass the BUSY or DECLINED codes to end point if that is experienced. I want the user to wait on MOH for example until the call is connected or until all routes are exhausted and then give him a BUSY. What would dialplan for something like this look like in Asterisk 1.8? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121204/b1949573/attachment.htm>
Pete Mundy
2012-Dec-05 18:38 UTC
[asterisk-users] How to check channel status and move on silently?
Dear list (FTPer, think I finally spotted one I can help with!)> I have 10 different routes with few different providers. When I place an international call, I would like the system to try all those routes and place the call through whichever possible. If there is any message but an ANSWER the system should move on to next route. I know this is not the best strategy but there are so many bad routes now-a-days that it's becoming a headache.I'd recommend looking at the ${DIALSTATUS} Asterisk variable and wrapping up some 'Dial' and 'Goto' applications in a macro that calls the first provider then looks at the returned ${DIALSTATUS} to make the logic decision of where to go from there. Then use the new macro everywhere where you would have previously used a single upstream dial. Further reading references (the second one actually containing an example that might be a useful starting point for you to work from): http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS http://www.voip-info.org/wiki/view/Asterisk+cmd+Goto Hope this helps! Pete Mundy -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4358 bytes Desc: not available URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121206/a7f9da5a/attachment.bin>