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

[David Gomillion]

On 5/15/07, J. David Bavousett <davidb at alc.org> wrote:
>
> 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


I recommend that you put in a group, like group=2

context=external
> signalling=fxs_ks
> channel => 5-8
> --------
>
> A snippet from /etc/asterisk/extensions.conf:
> --------
> [internal]
> ignorepat => 9


if you put in the group, you can dial out via:
exten => _9NXXXXXX,1,Dial(Zap/g2/${EXTEN:1}) to start with the lowest
available channel, or
                                  Dial(ZAP/G2/${EXTEN:1}) to start with the
highest available channel.

This will let you make more than one outgoing call at a time.

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.


Dial tone is generated by the SIP phone. You'll need to configure it
directly on whatever SIP device you're using. Now, if your analog phones
(like on ports 1-4) stop dial tone, you might need to be concerned.

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.


This is "normal" when dealing with POTS lines. You can try to get disconnect
supervision, try to trick zaptel into guessing what the state of the line
is, but in my experience, it just comes with the territory. Disconnect
supervision is, by far, the best solution, but most telcos stick their
fingers in their ears when it's requested...

That's one of the main reasons we use PRI where it makes sense, and have
people hang up the phones where it doesn't.

 <snip>

>
> 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'm not sure on this one. It could be a bad line, the line may not be fully
reset from the previous call, or something completely different.
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