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<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=945481116-19122005>The only other thing I can think of is that your
contexts etc... need checked. </SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=945481116-19122005></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=945481116-19122005>It would be very helpful to know if calls can come into
the system from the PBX, that would be the only way to know the span is alive
and well truely. Once you know that then its down to the contexts and
configs...</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=945481116-19122005></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=945481116-19122005></SPAN></FONT> </DIV>
<DIV> </DIV>
<DIV align=left><FONT face=Arial size=2></FONT> </DIV>
<DIV align=left><FONT face=Arial size=2>Jonathan O'Connor</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Senior System Administrator</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Inoveris LLC</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Direct Line (614)
791-3742</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Fax (614) 791-3748</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Helpdesk <FONT
color=#0000ff>866-456-1566</FONT></FONT></DIV>
<DIV align=left> </DIV>
<DIV align=left><FONT face=Arial size=2></FONT> </DIV>
<DIV> </DIV><BR>
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style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Michael
Sampson<BR><B>Sent:</B> Monday, December 19, 2005 11:05 AM<BR><B>To:</B>
Asterisk Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[Asterisk-Users] Can't call out on ZAP channel - need
help<BR></FONT><BR></DIV>
<DIV></DIV><TT>My other pbx vendor told me they supported pretty much all of
the switchtypes and that the system would automatically detect the correct
one. I've tried Qsig and National and both seem to bring the span up fine.
<BR><BR>I just switched to </TT><SPAN
class=060442015-19122005><TT>span=1,0,0,esf,b8zs to have asterisk provide the
timing. That didn't change any of the errors I'm getting. So I changed the
switchtype to national just to be sure, and it still didn't fix anything.
Everything seems to indicate that the span is up and running
fine.</TT><BR><BR>Any more ideas?<BR></SPAN><PRE class=moz-signature cols="72">Michael Sampson
Information Systems Manager
Customer Contact Services
<A class=moz-txt-link-abbreviated href="mailto:msampson@yourccsteam.com">msampson@yourccsteam.com</A>
952-936-4000</PRE><BR><BR>O'Connor, Jonathan wrote:
<BLOCKQUOTE
cite=mid2942C1A0B909B24CB7330E4FC6B7D2AE0E81FE4C@exchange01.metatec.com
type="cite">
<META content="MSHTML 6.00.2900.2627" name=GENERATOR>
<DIV dir=ltr align=left><SPAN class=060442015-19122005><FONT face=Arial
color=#0000ff size=2>The parameter in zaptel.conf that sets up timing etc
is:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005><FONT
face=Arial>span=1,1,0,esf,b8zs<BR></FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005><FONT face=Arial
color=#0000ff size=2>The first <STRONG>1</STRONG> means this is span
1. The second one defines the timing of the link. For asterisk
to provide the timing use <STRONG>0</STRONG> instead. For instance my
Asterisk box, hooked directly to my Avaya G3 uses:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN
class=060442015-19122005>span=1,0,0,esf,b8zs<BR></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005><FONT face=Arial
color=#0000ff size=2>Also,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005>switchtype=qsig
</SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005><FONT face=Arial
color=#0000ff size=2>This is something I have never personally got working
to any useful amount with our Definity. I use</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005><FONT
face=Arial>switchtype=national </FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005><FONT face=Arial
color=#0000ff size=2>It doesnt have some of the features of qsig, but will
get you going if the PBX is setup to use a standard National ISDN 2
switch.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005><FONT face=Arial
color=#0000ff size=2>You will I beleive need to shut down asterisk and then
run ztcfg -vvvv if you make these changes, then restart
asterisk.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005><FONT face=Arial
color=#0000ff size=2>signalling=pri_net merely makes the Asterisk box act
like the telco, as far as its signaling is concerned, quite normal when
hooked to a legecy pbx.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005><FONT face=Arial
color=#0000ff size=2>Hope this helps, am no expert, just going on what I got
mine running with :)</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005><FONT face=Arial
color=#0000ff size=2>-Jonathan</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=060442015-19122005></SPAN> </DIV>
<DIV> </DIV>
<DIV align=left> </DIV>
<DIV align=left><FONT face=Arial size=2>Jonathan O'Connor</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Senior System
Administrator</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Inoveris LLC</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Direct Line (614)
791-3742</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Fax (614) 791-3748</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Helpdesk <FONT
color=#0000ff>866-456-1566</FONT></FONT></DIV>
<DIV align=left> </DIV>
<DIV align=left> </DIV>
<DIV> </DIV><BR>
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style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(0,0,255) 2px solid; MARGIN-RIGHT: 0px">
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<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> <A class=moz-txt-link-abbreviated
href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</A>
[<A class=moz-txt-link-freetext
href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</A>]
<B>On Behalf Of </B>Michael Sampson<BR><B>Sent:</B> Monday, December 19,
2005 10:17 AM<BR><B>To:</B> Asterisk Users Mailing List - Non-Commercial
Discussion<BR><B>Subject:</B> Re: [Asterisk-Users] Can't call out on ZAP
channel - need help<BR></FONT><BR></DIV><TT>Yeah, the zttool program shows
the PRI as having No Alarms. It is an Infinity system by Amtelco. I
haven't actually tried making a call from the other pbx, but I did have my
vendor (Amtelco) look at it and they verified that the span was up and
everything was working correctly. The asterisk system is set to
signalling=pri_net which I assumed meant that the asterisk box would be
handling the timing. <BR><BR>Here is the output from "pri show span
1"<BR><BR>asterisk1*CLI> pri show span 1<BR>Primary D-channel:
24<BR>Status: Provisioned, Up, Active<BR>Switchtype: Q.SIG switch<BR>Type:
Network<BR>Window Length: 0/7<BR>Sentrej: 0<BR>SolicitFbit: 0<BR>Retrans:
0<BR>Busy: 0<BR>Overlap Dial: 0<BR>T200 Timer: 1000<BR>T203 Timer:
10000<BR>T305 Timer: 30000<BR>T308 Timer: 4000<BR>T313 Timer:
4000<BR><BR></TT><PRE class=moz-signature cols="72">Michael Sampson
Information Systems Manager
Customer Contact Services
<A class=moz-txt-link-abbreviated href="mailto:msampson@yourccsteam.com">msampson@yourccsteam.com</A>
952-936-4000</PRE><BR><BR>O'Connor, Jonathan wrote:
<BLOCKQUOTE
cite=mid2942C1A0B909B24CB7330E4FC6B7D2AE0E81FE4A@exchange01.metatec.com
type="cite"><PRE wrap="">Michael,
Does the zttool program show the PRI as working correctly?
Can the PBX push calls into the Asterisk system?
Also, what type of PBX is it, and is it providing the clock etc.. For
the T1 connection?
-Jonathan
Jonathan O'Connor
Senior System Administrator
Inoveris LLC
Direct Line (614) 791-3742
Fax (614) 791-3748
Helpdesk 866-456-1566
</PRE>
<BLOCKQUOTE type="cite"><PRE wrap="">-----Original Message-----
From: <A class=moz-txt-link-abbreviated href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</A>
[<A class=moz-txt-link-freetext href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</A>] On Behalf Of
Michael Sampson
Sent: Monday, December 19, 2005 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Can't call out on ZAP channel - need help
I'm trying to connect to another PBX via an T-1 interface. I
have a T100P card.
On the CLI I get the error "Everyone is busy/congested at
this time (1:0/0/1)" When I try to dial out of the T-1 line
from an SIP softphone.
I have posted this question a few times here and at the
asterisk forum, but can't get anyone to respond. I've seen
other people on forums with the same problem but no one has
ever given much of a solution. Does someone at least know
what the next step in debugging this problem would be.
In the file /var/log/asterisk/full I get the error "Unable to
create channel of type 'ZAP'"
Here are my configs.
Zapata.conf
------------------
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-pstn
;signalling=fxs_ks
signalling=pri_net ; pri_cpe= PRI slave ; pri_net = PRI
master switchtype=qsig pridialplan=local resetinterval=never
;rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO
lines ; ;usedistinctiveringdetection=yes callerid=asreceived
usecallerid=yes hidecallerid=no callwaiting=yes
usecallingpres=yes callwaitingcallerid=yes
threewaycalling=yes transfer=yes cancallforward=yes
callreturn=yes echocancel=yes echocancelwhenbridged=yes
echotraining=400 rxgain=0.0 txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;Include genzaptelconf configs
#include zapata-auto.conf
;Include AMP configs
#include zapata_additional.conf
channel => 1-23
-------------------------
Zaptel.conf
-------------------------
# Autogenerated by /usr/local/sbin/genzaptelconf -- do not
hand edit # Zaptel Configuration File # # This file is parsed
by the Zaptel Configurator, ztcfg #
# It must be in the module loading order
# Span 1: WCT1/0 "Digium Wildcard T100P T1/PRI Card 0"
# channel 1, WCT1, unhandled for now
# channel 2, WCT1, unhandled for now
# channel 3, WCT1, unhandled for now
# channel 4, WCT1, unhandled for now
# channel 5, WCT1, unhandled for now
# channel 6, WCT1, unhandled for now
# channel 7, WCT1, unhandled for now
# channel 8, WCT1, unhandled for now
# channel 9, WCT1, unhandled for now
# channel 10, WCT1, unhandled for now
# channel 11, WCT1, unhandled for now
# channel 12, WCT1, unhandled for now
# channel 13, WCT1, unhandled for now
# channel 14, WCT1, unhandled for now
# channel 15, WCT1, unhandled for now
# channel 16, WCT1, unhandled for now
# channel 17, WCT1, unhandled for now
# channel 18, WCT1, unhandled for now
# channel 19, WCT1, unhandled for now
# channel 20, WCT1, unhandled for now
# channel 21, WCT1, unhandled for now
# channel 22, WCT1, unhandled for now
# channel 23, WCT1, unhandled for now
# channel 24, WCT1, unhandled for now
# Global data
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
#fxsks=1
loadzone = us
defaultzone = us
-----------------------------
--
Michael Sampson
Information Systems Manager
Customer Contact Services
<A class=moz-txt-link-abbreviated href="mailto:msampson@yourccsteam.com">msampson@yourccsteam.com</A>
952-936-4000
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