Chris Blunt
2007-Jul-30 15:09 UTC
[asterisk-users] Dial plan question: PSTN via Linksys SPA3102 then IAX if busy?
Hi All, In our small office calls to the PSTN are currently sent via Asterisk and a Linksys SPA3102 (1 x FXO and 1 x FXS): SIP Phone --> Asterisk --> Linksys SPA3102 --> PSTN If the PSTN is in use on SPA3102 I need a way to get the call to then route out over IAX termination. SIP Phone --> Asterisk--> Linksys SPA3102 --> PSTN (In Use) --> Use IAX Can any one help me with some dial plan logic for this; I'm confused as to the best way around this? Thanks in advance Chris -- Chris Blunt Entropy IT Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070730/e75c8d66/attachment.htm
Jared Smith
2007-Jul-31 00:52 UTC
[asterisk-users] Dial plan question: PSTN via Linksys SPA3102 then IAX if busy?
On Mon, 2007-07-30 at 16:09 +0100, Chris Blunt wrote:> If the PSTN is in use on SPA3102 I need a way to get the call to then > route out over IAX termination.Usually, the best way to accomplish this is to send a call to your Linksys ATA by using the Dial application from the dialplan, and then looking at the result that gets set in the DIALSTATUS variable. For example, you could try something like this: exten => 123,1,Dial(SIP/linksys/5551212,30) exten => 123,n,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?try-iax) exten => 123,n,Busy(3) exten => 123,n,Hangup() exten => 123,n(try-iax),Dial(IAX2/my_iax_peer/5551212,30) Obviously my example isn't that robust... it's simply meant to illustrate the idea. (It depends on the SPA3102 returning a status code that maps to CONGESTION if it's already in use... I don't have an SPA3102, so I can't tell you how it actually performs.) -- Jared Smith Community Relations Manager Digium, Inc.