Tim,
I've tested similar dialplan on my home-server and it works perfectly.
(same setting, slightly different extensions) but same idea:
exten => 418,1,Dial(SIP/55,15,trw)
exten => 418,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
exten => 418,n(line2),Dial(SIP/218,15,rw)
exten => 418,n(vmail),Voicemail(55)
exten => 418,n,Voicemail(55)
exten => 418,n,Hangup()
I think the reason the below dialpolan IS NOT WORKING is that I'm connecting
(dialing) remote asterisk extension.
--------not working calling remote asterisk----------
exten => 4,1,Dial(${FD_L1},25,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
exten => 4,n(line2),Dial(${FD_L2},20,rw)
exten => 4,n(vmail),Voicemail(4)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
-----end not working calling remote asterisk---------
I have two Asterisk server connected/registered over IAX and that error:
"...exited non-zero on..."
eg.
-- SIP/54-00000006 is ringing
== Spawn extension (extensions, 4, 3) exited non-zero on
'IAX2/home_server-424'
I'm not the only one with this problem, this guy has the same problem as me:
http://lists.digium.com/pipermail/asterisk-users/2006-January/135612.html
--
Thelma
On 05/08/2017 06:58 PM, Tim S wrote:
> So, good, we're on the same page so far I think.
>
> As I last stated, the original code suggestion would be what you want to
> do for the serial phone ring-down (hunt), now you just need to figure
> out why your Line_2 phone is answering and then hanging up immediately
> (or why Asterisk thinks it is).
>
> I'd recommend sniffing the network traffic with Wire Shark and turning
> on some of the debug options in Asterisk to hunt down if it's the phone
> or an Asterisk quirk that is tripping up the system. We'll need more
> debug and error text to go any further with the Line_2 problem, unless
> someone much better than me can chime in with an idea... I presume
> you've already done the simple stuff like make sure your network is
> solid and that the phone firmware is up to date and stable.
>
> I'll also take a moment as an aside to suggest that you move away from
> numerical device and user names for SIP and move to text based names
> which have local meaning. The numerical names are easy to be hacked, as
> bad-guys scripts easily walk the possibilities sequentially. I find it
> also helps to use extension names in the dial plan that have meaning so
> that I can keep track of them. When a user calls an extension, the
> number they enter can feature a "Goto" with a text entry in the dial
> plan. This makes it harder for those at a phone to go places in your
> phone system they shouldn't.
>
> -Tim
>
> On Mon, May 8, 2017 at 4:51 PM, <the...@sys-concept.com
> <mailto:the...@sys-concept.com>> wrote:
>
> 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
>
> --
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