[asterisk-users] Call does not go to voicemail
thelma at sys-concept.com
thelma at sys-concept.com
Mon May 8 18:51:37 CDT 2017
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 at extensions:2] GotoIf("IAX2/home_server-435", "0?line2:") in new stack
-- Executing [4 at 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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