<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
<tt>As it turns out I can dial from the Infinity PBX into the Asterisk
box. So it must be something to do with contexts or configs I guess. <br>
<br>
So when I set up the ZAP trunk in AMP it automatically did it as
ZAP/g0. Well I just assumed that was the first one. Most spans
numbering that I deal with starts with 0. I changed it to ZAP/g1 and
everything seems to be working now. I can dial both ways. Thanks for
you help, I would have been stuck thinking there was something wrong
with the span setup otherwise.<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="mid2942C1A0B909B24CB7330E4FC6B7D2AE0E81FE4E@exchange01.metatec.com"
type="cite">
<meta http-equiv="Content-Type" content="text/html; ">
<meta content="MSHTML 6.00.2900.2627" name="GENERATOR">
<div align="left" dir="ltr"><font color="#0000ff" face="Arial"
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 align="left" dir="ltr"><font color="#0000ff" face="Arial"
size="2"><span class="945481116-19122005"></span></font> </div>
<div align="left" dir="ltr"><font color="#0000ff" face="Arial"
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 align="left" dir="ltr"><font color="#0000ff" face="Arial"
size="2"><span class="945481116-19122005"></span></font> </div>
<div align="left" dir="ltr"><font color="#0000ff" face="Arial"
size="2"><span class="945481116-19122005"></span></font> </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>
<blockquote
style="border-left: 2px solid rgb(0, 0, 255); padding-left: 5px; margin-left: 5px; margin-right: 0px;"
dir="ltr">
<div class="OutlookMessageHeader" align="left" dir="ltr"
lang="en-us">
<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 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>
<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 align="left" dir="ltr"><span class="060442015-19122005"><font
color="#0000ff" face="Arial" size="2">The parameter in zaptel.conf
that sets up timing etc is:</font></span></div>
<div align="left" dir="ltr"><span class="060442015-19122005"></span> </div>
<div align="left" dir="ltr"><span class="060442015-19122005"><font
face="Arial">span=1,1,0,esf,b8zs<br>
</font></span></div>
<div align="left" dir="ltr"><span class="060442015-19122005"><font
color="#0000ff" face="Arial" 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 align="left" dir="ltr"><span class="060442015-19122005"></span> </div>
<div align="left" dir="ltr"><span class="060442015-19122005">span=1,0,0,esf,b8zs<br>
</span></div>
<div align="left" dir="ltr"><span class="060442015-19122005"><font
color="#0000ff" face="Arial" size="2">Also,</font></span></div>
<div align="left" dir="ltr"><span class="060442015-19122005"></span> </div>
<div align="left" dir="ltr"><span class="060442015-19122005">switchtype=qsig
</span></div>
<div align="left" dir="ltr"><span class="060442015-19122005"></span> </div>
<div align="left" dir="ltr"><span class="060442015-19122005"></span> </div>
<div align="left" dir="ltr"><span class="060442015-19122005"><font
color="#0000ff" face="Arial" size="2">This is something I have never
personally got working to any useful amount with our Definity. I use</font></span></div>
<div align="left" dir="ltr"><span class="060442015-19122005"></span> </div>
<div align="left" dir="ltr"><span class="060442015-19122005"><font
face="Arial">switchtype=national </font></span></div>
<div align="left" dir="ltr"><span class="060442015-19122005"></span> </div>
<div align="left" dir="ltr"><span class="060442015-19122005"><font
color="#0000ff" face="Arial" 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 align="left" dir="ltr"><span class="060442015-19122005"></span> </div>
<div align="left" dir="ltr"><span class="060442015-19122005"><font
color="#0000ff" face="Arial" 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 align="left" dir="ltr"><span class="060442015-19122005"></span> </div>
<div align="left" dir="ltr"><span class="060442015-19122005"></span> </div>
<div align="left" dir="ltr"><span class="060442015-19122005"><font
color="#0000ff" face="Arial" 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 align="left" dir="ltr"><span class="060442015-19122005"></span> </div>
<div align="left" dir="ltr"><span class="060442015-19122005"><font
color="#0000ff" face="Arial" size="2">Hope this helps, am no expert,
just going on what I got mine running with :)</font></span></div>
<div align="left" dir="ltr"><span class="060442015-19122005"></span> </div>
<div align="left" dir="ltr"><span class="060442015-19122005"></span> </div>
<div align="left" dir="ltr"><span class="060442015-19122005"><font
color="#0000ff" face="Arial" size="2">-Jonathan</font></span></div>
<div align="left" dir="ltr"><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>
<blockquote
style="border-left: 2px solid rgb(0, 0, 255); padding-left: 5px; margin-left: 5px; margin-right: 0px;"
dir="ltr">
<div class="OutlookMessageHeader" align="left" dir="ltr"
lang="en-us">
<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
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
<pre wrap=""><!---->_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
</blockquote>
<pre wrap=""><hr size="4" width="90%">
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
</blockquote>
<pre wrap="">
<hr size="4" width="90%">
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
</body>
</html>