Fwd: [Asterisk-Users] Local Outbound Calls on PRI

[Paul Oster]

---------- Forwarded message ----------
From: Paul Oster <paulmoster at gmail.com>
Date: Mon, 27 Sep 2004 09:39:05 -0500
Subject: Re: [Asterisk-Users] Local Outbound Calls on PRI
To: Steve Maroney <steve at stevenet.net>

Console Output from Long Distance Call

    -- Executing Dial("SIP/7013560112-8780", "Zap/g1/17012402815") in new stack
    -- Called g1/17012402815
    -- Zap/1-1 is ringing
    -- Hungup 'Zap/1-1'

Console Output from Local Call
    -- Executing Dial("SIP/7013560112-206b", "Zap/g1/3566150") in new stack
    -- Called g1/3566150
    -- Hungup 'Zap/1-1'

Both cases Hangup line does not occur untill I physically hang up the SIP phone.
Long Distance Call rings through (its my cell phone) with the correct
CID information for the line.

Local Call generates a "fast-busy" or re-order tone immediately (less
than 1 second) after the Called line.

exten => 411,1,Dial(Zap/g1/${EXTEN})
exten => 411,2,Busy

exten => 911,1,Dial(Zap/g1/${EXTEN})
exten => 911,2,Busy

exten => _NXXXXXX,1,Dial(Zap/g1/${EXTEN})
exten => _NXXXXXX,2,Congestion

exten => _NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN})
exten => _NXXNXXXXXX,2,Congestion

Above is the trunks section of my dialplan.

Zap/g1 is channesl 1-23 on the first (and only) PRI on my Quad T1 card.

Here is my zaptel.conf
loadzone = us
defaultzone=us
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
fxoks=97-98

Here is my zapata.conf

switchtype=national
pridialplan=national
signalling=pri_cpe
context=incomingcall
group=1
channel=>1-23

signalling=fxo_ks
context=employee
channel => 97-98

The pridialplan is set at the request of the telco, the same behavior
was observed with the pridial plan missing from the config as would be
normal for most PRI's.

Dialing directory assistance as 411 also works in addition to the LD
calls that have been working.

Here is the sip.conf file currently in use (reduced to 1 entry during testing)
[7013560112]
accountcode=pauloster
type=friend
context=phonecustomers
username=7013560112
secret=<deleted>
host=dynamic
dtmfmode=inband
qualify=yes
reinvite=no
mailbox=7013560112
callerid=7013560112

Any suggestions would be greatly appreciated.




On Fri, 24 Sep 2004 14:45:02 -0500 (CDT), Steve Maroney
<steve at stevenet.net> wrote:
> Your extensions.conf and console output might help us out quite a bit.
>
> Thank you,
> Steve Maroney
>
>
>
> On Fri, 24 Sep 2004, Paul Oster wrote:
>
> > I'm in the process of turning up a PRI in one of my markets and have
> > run into a problem I have never seen before.  I am unable to place a
> > local outgoing call.  Long Distance over the same PRI works fine.
> >
> > When I attempt to place a local call using the PRI I see Asterisk
> > attempt to dial, and am greeted with a busy signal.  This signal
> > appears to originate on the telco's switch.
> >
> > I have had a central office tech from the CLEC insert a monitor in the
> > distribution point on the switch and observe call flow.  According to
> > the tech call flow appears proper, and he was able to tell me the
> > number I was calling from and the number I attempted to dial.
> >
> > He then placed a PRI test-set at the distribution point in the switch
> > and successfully made and terminated a variety of calls from that
> > point.
> >
> > He then took the PRI test set out to my physical location and did the
> > same test, made and received local and long distance calls on one of
> > my trunks.
> >
> > After thinking about this last night I decided to re-update my
> > Asterisk installation to the latest bleeding edge CVS version and
> > re-test from my test extension with the exact same results.  I
> > successfully make long distance calls, successfully receive any calls,
> > but the local calls originated from the SIP phone (SNOM200 and
> > Mediatrix2102) fail with a busy signal that seems to originate from
> > the CLEC's switch.
> >
> > Any suggestions?
> >
> > Thanks in advance.
> >
> > Paul
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>