[Asterisk-Users] Asterisk@home questions
Sascha Ferley
Sascha.Ferley at infineon.net
Wed Apr 27 23:22:00 MST 2005
Hi
I am currently running asterisk at home version 0.9 and have a few questions,
which i hope someone on this list might be able to answer.
1) I am trying to setup incomming fax support, but however i never manage
to receive the faxes, getting a signal 15. As per handbook, there isn't
too much underlaying documentation, which one could reference to debug the
problems. Here is a log:
pr 24 03:01:01 NOTICE[1053]: Got event 2 (Ring/Answered)...
Apr 24 03:01:02 DEBUG[1053]: Expression is '0'
Apr 24 03:01:02 VERBOSE[1053]: -- Executing GotoIf("40mZap/3-1", "0?from-pstn-reghours|s|1:") in new stack
Apr 24 03:01:02 DEBUG[1053]: Not taking any branch
Apr 24 03:01:02 DEBUG[1053]: Expression is '0'
Apr 24 03:01:02 VERBOSE[1053]: -- Executing GotoIf("40mZap/3-1", "0?from-pstn-afthours|s|1:") in new stack
Apr 24 03:01:02 DEBUG[1053]: Not taking any branch
Apr 24 03:01:02 VERBOSE[1053]: -- Executing GotoIfTime(";35;40mZap/3-1", "*|*|*|*?from-pstn-reghours|s|1:") in new stack
Apr 24 03:01:02 VERBOSE[1053]: -- Goto (from-pstn-reghours,s,1)
Apr 24 03:01:02 DEBUG[1053]: Expression is '0'
Apr 24 03:01:02 VERBOSE[1053]: -- Executing GotoIf("40mZap/3-1", "0?from-pstn-reghours-nofax|s|1:2") in new stack
Apr 24 03:01:02 VERBOSE[1053]: -- Goto (from-pstn-reghours,s,2)
Apr 24 03:01:02 VERBOSE[1053]: -- Executing Answer("40mZap/3-1", "") in new stack
Apr 24 03:01:02 DEBUG[1053]: Took Zap/3-1 off hook
Apr 24 03:01:02 DEBUG[1053]: Enabled echo cancellation on channel 3
Apr 24 03:01:02 DEBUG[1053]: Engaged echo training on channel 3
Apr 24 03:01:02 VERBOSE[1053]: -- Executing Wait("mZap/3-1", "1") in new stack
Apr 24 03:01:03 VERBOSE[1053]: -- Executing SetVar("40mZap/3-1", "intype=GRP-1") in new stack
Apr 24 03:01:03 VERBOSE[1053]: -- Executing Cut("Zap/3-1", "intype=intype|-|1") in new stack
Apr 24 03:01:03 DEBUG[1053]: Expression is '0'
Apr 24 03:01:03 VERBOSE[1053]: -- Executing GotoIf("40mZap/3-1", "0?7:9") in new stack
Apr 24 03:01:03 VERBOSE[1053]: -- Goto (from-pstn-reghours,s,9)
Apr 24 03:01:03 DEBUG[1053]: Expression is '1'
Apr 24 03:01:03 VERBOSE[1053]: -- Executing GotoIf("40mZap/3-1", "1?10:12") in new stack
Apr 24 03:01:03 VERBOSE[1053]: -- Goto (from-pstn-reghours,s,10)
Apr 24 03:01:03 VERBOSE[1053]: -- Executing Wait("mZap/3-1", "3") in new stack
Apr 24 03:01:04 DEBUG[1053]: DTMF digit: f on Zap/3-1
Apr 24 03:01:04 VERBOSE[1053]: -- Redirecting Zap/3-1 to fax extension
Apr 24 03:01:04 VERBOSE[1053]: == Spawn extension (from-pstn-reghours, fax, 0) exited non-zero on 'Zap/3-1'
Apr 24 03:01:04 VERBOSE[1053]: -- Executing Goto("mZap/3-1", "ext-fax|in_fax|1") in new stack
Apr 24 03:01:04 VERBOSE[1053]: -- Goto (ext-fax,in_fax,1)
Apr 24 03:01:04 DEBUG[1053]: Expression is '1'
Apr 24 03:01:04 VERBOSE[1053]: -- Executing GotoIf("40mZap/3-1", "1?2:analog_fax|1") in new stack
Apr 24 03:01:04 VERBOSE[1053]: -- Goto (ext-fax,in_fax,2)
Apr 24 03:01:04 VERBOSE[1053]: -- Executing Macro("0mZap/3-1", "faxreceive") in new stack
Apr 24 03:01:04 VERBOSE[1053]: -- Executing SetVar("40mZap/3-1", "FAXFILE=/var/spool/asterisk/fax/1114333259.710.tif") in new stack
Apr 24 03:01:04 VERBOSE[1053]: -- Executing SetVar("40mZap/3-1", "EMAILADDR=fax at mydomain.net") in new stack
Apr 24 03:01:04 VERBOSE[1053]: -- Executing RxFAX("0mZap/3-1", "/var/spool/asterisk/fax/1114333259.710.tif0m") in new stack
Apr 24 03:02:15 DEBUG[1053]: Exception on 15, channel 3
Apr 24 03:02:15 DEBUG[1053]: Got event On hook(1) on channel 3 (index 0)
Apr 24 03:02:15 DEBUG[1053]: disabled echo cancellation on channel 3
Apr 24 03:02:15 DEBUG[1053]: Got hangup
Apr 24 03:02:15 DEBUG[1053]: Extension s, priority 3 returned normally even though call was hung up
Apr 24 03:02:15 DEBUG[1053]: Extension in_fax, priority 2 returned normally even though call was hung up
Apr 24 03:02:15 VERBOSE[1053]: -- Executing Hangup("40mZap/3-1", "") in new stack
Apr 24 03:02:15 VERBOSE[1053]: == Spawn extension (ext-fax, h, 1) exited non-zero on 'Zap/3-1'
Thus as one sees it seems i get a exception. I did do the asterisk pdf
install. Is there any way to get better debugging info? Also any way to
configure asterisk to keep the tiff files around and not delete them when
they are sent?
2) The next question i have is how do i manipulate a physical line via
asterisk on a sip client. Basically i am subscribed to some services via
the telco like call waiting / call forwarding, where as using the *70 on a
sip will just do a call forwarding on asterisk and not send that signal
out to the telco. Also trying to manipulate the call waiting signal on the
line, as when i am on the sip phone i can hear a call come in, but can't
seem to instruct asterisk to send the switch voice channel signal to the
telco. Any ideas on how to do this? Not in the documentation from what i
can read.
Please let me know
Thanks
Sascha
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