[asterisk-users] Outside lines are just not happening...

J. David Bavousett davidb at alc.org
Tue May 15 09:47:23 MST 2007


Two problems, possibly related:

Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.

Here are the config files:

/etc/zaptel.conf:
--------
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8

loadzone        = us
defaultzone     = us
--------
/etc/asterisk/zapata.conf:
--------
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
canpark=yes
rxgain=0.0
txgain=0.0

context=internal
signalling=fxo_ks
channel => 1-4

context=external
signalling=fxs_ks
channel => 5-8
--------

A snippet from /etc/asterisk/extensions.conf:
--------
[internal]
ignorepat => 9
exten => _9NXXXXXX,1,Dial(Zap/5/${EXTEN:1})
exten => _9NXXXXXX,2,Congestion()
exten => _9NXXXXXX,102,Congestion()
--------

The SIP phone is also in the internal context, and other things below
that in the context work just fine on the internal network.

I don't know if it's relevant or not, but dialtone stops after I press
9, which is not what I was led to believe would happen with the
ignorepat directive.

Problem A:  Dialing in.  If I call from my cell, the FXO picks right up,
and sends me to the voice menu that I have at the top of the [external]
context.  So far so good, but if the SIP that I get in touch with hangs
up, the FXO stays off-hook for more than a minute before dropping the
POTS line.  If I pick that SIP phone back up, and dial an outside
number, I can reconnect to the "dangling" call, which will hear the
tones after the 9...  The outside caller will finally get dropped after
about a minute of waiting.

Here's a transcript from the CLI:
(I pick up a phone not on the switch, and call the FXO:
    -- Starting simple switch on 'Zap/5-1'
    -- Executing Answer("Zap/5-1", "") in new stack
    -- Executing GotoIfTime("Zap/5-1",
"07:30-16:30|mon-fri|*|*?open|s|1") in new stack
    -- Goto (open,s,1)
    -- Executing DigitTimeout("Zap/5-1", "5") in new stack
    -- Set Digit Timeout to 5
    -- Executing ResponseTimeout("Zap/5-1", "10") in new stack
    -- Set Response Timeout to 10
    -- Executing BackGround("Zap/5-1", "alc01") in new stack
    -- Playing 'alc01' (language 'en')
  == CDR updated on Zap/5-1
    -- Executing Macro("Zap/5-1", "stdext|102|SIP/102") in new stack
    -- Executing Dial("Zap/5-1", "SIP/102|20") in new stack
    -- Called 102
(The SIP phone begins ringing)
    -- SIP/102-08184fa8 is ringing
    -- SIP/102-08184fa8 answered Zap/5-1
(Hang up SIP)
  == Spawn extension (macro-stdext, s, 1) exited non-zero on 'Zap/5-1'
in macro 'stdext'
  == Spawn extension (macro-stdext, s, 1) exited non-zero on 'Zap/5-1'
    -- Hungup 'Zap/5-1'
(Pick up SIP, dial 96653674)
    -- Executing Dial("SIP/102-08184808", "Zap/5/6653674") in new stack
    -- Called 5/6653674
    -- Zap/5-1 answered SIP/102-08184808
(I heard the tones on the "outside" phone, which is still off-hook)
(hung up SIP)
    -- Hungup 'Zap/5-1'
  == Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-08184808'
(one more time, pick up SIP)
    -- Executing Dial("SIP/102-08184808", "Zap/5/6653674") in new stack
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Congestion("SIP/102-08184808", "") in new stack
(Got fast-busy, so hung up SIP)
  == Spawn extension (internal, 96653674, 2) exited non-zero on
'SIP/102-08184808'
(about now, the outside line went back to dialtone.)

Sometimes, I can repeat that pick-back-up trick three or four times.

Problem B:  Dialing out.  From the SIP phone, if I dial out, here's the
transcript:

    -- Executing Dial("SIP/102-081854e0", "Zap/5/6653674") in new stack
    -- Called 5/6653674
    -- Zap/5-1 answered SIP/102-081854e0
    -- Hungup 'Zap/5-1'
  == Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-081854e0'

Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call.  The SIP
phone hangs up after about 5 seconds.  But not always.  The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
own, and the cell never reports a missed call.

I know this has been long, and wordy...hope someone can help.  We're
newbs around here, and trying to get things working.  My boss is *very*
impressed with the menus and such I've got set up and working, and we've
used soft phones via VPN and it works great...now we just need our
outside lines working!

Thanks a million!

J. David Bavousett
System Administrator
Abilene Library Consortium



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