On 05/08/2017 04:37 PM, Tim S wrote: > The "error" I was talking about was in your log: > > "...== Spawn extension (extensions, 4, 3) exited non-zero on > 'IAX2/home_server-6364'..." > > The call terminated here in a error which prevented the dialplan from > continuing. Something there is broken, my recommendation is to check > you registrations first inside asterisk: > >> sip show peers
"sip show peers" is showing FD_L2 (SIP/54 is registered) Name/username Host Dyn Forcerport Comedia ACL Port Status Description 12 (Unspecified) D No No 0 Unmonitored 4/4 10.10.0.8 D No No 5060 Unmonitored 54/54 10.10.0.15 D No No 5060 Unmonitored > Something wasn't "happy" about SIP/54 in your system when Asterisk tried > talking to it. > > So you tried this: > > "... > Even when I put: > exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2) > exten => 4,n(line2),Dial(${FD_L2},20,trw) > exten => 4,n(line2),Voicemail(4) > ..." > > What that will do is go to the first instance of "4,n(line2)", which is > the line that seems to be triggering the channel failure. If you have > the Asterisk console open, I'll bet you see it spew some errors when you > try that extension routine. > > Asterisk dial plans are a serial processes, the first line that Asterisk > comes across that meets the matching for a given extension and label is > what it will run first. What you have is two lines that will match both > extension and label - that's not really good form. > > My dial plan suggestion from last night would result in the functionality: > > Ring extension 4/Line_1, timeout 25 seconds --> if not busy then > voicemail, else ring extension 4/Line_2, timeout 20 seconds --> voicemail. > > > Again, I think you have two problems, and the bigger one is causing the > annoying unexpected behavior in your dial plan > > Try doing the extension 4 without the Line_1 and see what happens: > > "... > exten => 4,1,Dial(${FD_L2},20,trw) > exten => 4,n(vmail),Voicemail(4) > exten => 4,n,Hangup() > ..." I have tired the above plan with small change 4,n,Voicemail(4) (as there is no gotoif statement) So: exten => 4,1,Dial(${FD_L2},20,trw) exten => 4,n,Voicemail(4) exten => 4,n,Hangup() Line 2 is ring OK, and if nobody pickup the phone it goes to "Voicemail(4)" so this part is working; there were no errors on the command line. [snip] But I've tired it again, this dialplan) as before and you are correct something is wrong but command line is not showing any errors: exten => 4,1,Dial(${FD_L1},25,trw) exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2:) exten => 4,n(line2),Dial(${FD_L2},20,rw) exten => 4,n,Voicemail(4) exten => 4,n,Hangup() I've tried: exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2) exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2:) exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?:line2) And I get: -- Called SIP/4 -- SIP/4-00000306 is ringing -- Nobody picked up in 25000 ms -- Executing [4@extensions:2] GotoIf("IAX2/home_server-435", "0?line2:") in new stack -- Executing [4@extensions:3] Dial("IAX2/home_server-435", "SIP/54,20,rw") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/54 -- SIP/54-00000307 is ringing == Spawn extension (extensions, 4, 3) exited non-zero on 'IAX2/home_server-435' -- Hungup 'IAX2/home_server-435' So FD_L1 (exten: 4) is ringing for 25sec.; nobody pickup the phone and command line is showing it goes to: FD_L2 (SIP/54) -- SIP/54-00000307 is ringing but in reality FD_L2 (SIP/54) is not ringing at all, it should ring line_2 for 20sec and go to Voicemail but as soon as it prints line: -- SIP/54-00000307 is ringing it hangs up the phone. -- Thelma -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users