asterisk users - Apr 2005 - Asterisk@home questions

Hi

I am currently running asterisk@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@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

Hi!
I am new to this, and have been googling for hours to solve this issue.
 First question: How do I go about setting up my POTS line through a modem
that is in my server? I can also not get my "hardphone" to connect to
asterisk. The softphones can talk to each other and through a GoIAX VoIP
line.
 Second Question: (No direct how-to or answer online) How do I set up
Asterisk@Home to page through the soundcard in the server. I would like to
dial a code from a phone or selected phones (say 1234 as the code) and have
it play a beep and page the speakers.
 Thanks,
 Michael
 PS If this should not be here, would someone willing to help contact me
off-list? Thanks!
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Michael,
 
This is the list for Asterisk not Asterisk at Home.  That list list can
be found at the same place you downloaded the AAH software.
 
www.voip-info.org is the location of Asterisk Wiki.
 
Now.  On to your problems.
 
1. No one can help you with that problem since we don know what kind of
phone or what kind of modem.
you cannot use just any modem.  It has to be either a Digium TDM card or
a clone with the Tiger chipset.
Connection after that is a matter of FSO versus FXO.  Google for that if
explanation needed...
 
2. Go to Wiki I listed above and you can find how to setup paging.  Then
you just need to create an extension in the AAH setup that does what you
want.
 
And finally... If you are just getting started, you have about a 2 week
learning curve so immerse yourself in the Wiki and all will be well...
 
Good luck,
Wiley
 
 
 
 
 
 

________________________________

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael
Janofsky
Sent: Wednesday, September 28, 2005 4:01 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Asterisk@Home Questions


Hi!
I am new to this, and have been googling for hours to solve this issue.
 
First question: How do I go about setting up my POTS line through a
modem that is in my server?  I can also not get my "hardphone" to
connect to asterisk.  The softphones can talk to each other and through
a GoIAX VoIP line. 
 
Second Question: (No direct how-to or answer online) How do I set up
Asterisk@Home to page through the soundcard in the server.  I would like
to dial a code from a phone or selected phones (say 1234 as the code)
and have it play a beep and page the speakers. 
 
Thanks,
 
Michael
 
PS If this should not be here, would someone willing to help contact me
off-list?  Thanks!
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