asterisk users - Aug 2005 - TE110P flashing red/green when PRI connected

Hi 
 
I'm having difficulties getting up my TE110P (running as a E1) when I
connect it to the PRI. If I start the server with a loopback connector
everything seems fine and the led is green but when I connect it to the PRI the
flashing starts ....
 
I can't seem to find anything in the log that suggests what could be wrong 
 
.....Aug 8 11:33:00 DEBUG[1369]: Updated conferencing on 31, with 0 conference
users
Aug 8 11:33:00 VERBOSE[1369]: -- Registered channel 31, PRI Signalling
signalling
Aug 8 11:33:00 VERBOSE[1369]: -- Automatically generated pseudo channel 
Aug 8 11:33:00 VERBOSE[1369]: == Starting D-Channel on span 1 
Aug 8 11:33:00 VERBOSE[1369]: == Registered channel type 'Zap' (Zapata
Telephony Driver w/PRI)
Aug 8 11:33:00 VERBOSE[1369]: == Registered channel type 'Tor' (Zapata
Telephony Driver w/PRI)
Aug 8 11:33:00 VERBOSE[1369]: == Registered application 'CallingPres' 
Aug 8 11:33:00 VERBOSE[1369]: == Manager registered action ZapTransfer 
Aug 8 11:33:00 VERBOSE[1369]: == Manager registered action ZapHangup 
Aug 8 11:33:00 VERBOSE[1369]: == Manager registered action ZapDialOffhook 
Aug 8 11:33:00 VERBOSE[1369]: == Manager registered action ZapDNDon 
Aug 8 11:33:00 VERBOSE[1369]: == Manager registered action ZapDNDoff 
Aug 8 11:33:00 VERBOSE[1369]: == Manager registered action ZapShowChannels 
..... 
Is there anywhere else in the log you should look? 
 
Here is my zaptel.conf: 
 
# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" B8ZS/ESF  
 
# Global data 
span=1,0,0,ccs,hdb3,crc4 
bchan=1-15 
dchan=16 
bchan=17-31 
 
Note: If I specify loadzone se, asterisk won't start at all. Is there any
other loadzone I should use?
 
[trunkgroups] 
 
[channels] 
 
language=us  
context=from-pstn 
switchtype=euroisdn  
;If you connect to a hicom PBX set your ISDN Numbering Plan Identifier to
unknown.
pridialplan=unknown  
prilocaldialplan=unknown  
signalling=pri_cpe  
usecallerid=yes  
hidecallerid=no  
callwaiting=yes  
usecallingpres=yes  
callwaitingcallerid=yes  
threewaycalling=yes  
transfer=yes  
cancallforward=yes  
callreturn=yes  
echocancel=yes  
echocancelwhenbridged=yes  
rxgain=0.0  
txgain=0.0  
group=1  
callgroup=1  
pickupgroup=1  
immediate=no  
 
group = 1  
channel => 1-15  
channel => 17-31  
 
 
I'm located in Sweden and using Telenor as my provider. 
 
Anyone got any ideas?  
 
Thanks!

Hi,

I've got some problems with mythphone registering to asterisk and I 
can't seem to find the right combination to get dtmf working properly.

Anybody using it and have a sip.conf they want to share with me.

Thanks, Oliver

On Mon, 8 Aug 2005, Fredrik Lith?n wrote:
> I'm having difficulties getting up my TE110P (running as a E1) when I
> connect it to the PRI. If I start the server with a loopback connector
> everything seems fine and the led is green but when I connect it to the
> PRI the flashing starts ....
The command "zttool" may tell you what is wrong.

Peter

Peter,

Thanks for the reply!

Yes, I tried that but it sent me a bit offtrack as it reported "blue"
which I assumed was a clocksync problem, or at least, that was the info
I could find.

As it turned out, my provider didn't have error correction enabled so
after have endured painstaking task of getting hold of the right
person to talk to :) the problem was swiftly fixed.

Fredrik

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Peter Svensson
Sent: den 8 augusti 2005 16:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TE110P flashing red/green when PRI connected

On Mon, 8 Aug 2005, Fredrik Lith?n wrote:
> I'm having difficulties getting up my TE110P (running as a E1) when I 
> connect it to the PRI. If I start the server with a loopback connector 
> everything seems fine and the led is green but when I connect it to 
> the PRI the flashing starts ....
The command "zttool" may tell you what is wrong.

Peter


_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@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

Thanks for the clarification on "code blue"

Yup, the first thing I altered was to set span=1,1... as I thought that
the clocksync was the issue.

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Peter Svensson
Sent: den 9 augusti 2005 10:32
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TE110P flashing red/green when PRI connected

On Tue, 9 Aug 2005, Fredrik Lith?n wrote:
> Yes, I tried that but it sent me a bit offtrack as it reported
"blue"
> which I assumed was a clocksync problem, or at least, that was the 
> info I could find.
As far as I can tell zttool/zaptel uses the term BLue Alarm for the E1 term AIS
(Alarm Indication Signal) which is a layer 2 signal sent by someone in the
receive path (not necesarrily the PSTN end itself) indicating that no valid
incoming signal is present.

A bitstream is present at the receiver, though it is unframed and invalid (i.e.
the receiver is seeing a transmitter that does not quite know what to transmit).
This is different from a red alarm where there is no bitstream at all.

One common cause for the blue alarm is when the PSTN end shuts down an interface
that has many errors. This si common practice and is probably what happend when
there was a mismatch in the crc setting.
> As it turned out, my provider didn't have error correction enabled so 
> after have endured painstaking task of getting hold of the right 
> person to talk to :) the problem was swiftly fixed.
An additional point: in the original post you mentioned that your zaptel line
looked like
  span=1,0,0,ccs,hdb3,crc4
which should almost certainly be
  span=1,1,0,ccs,hdb3,crc4
since you want to synchronize your clock rate to that of the PSTN provider. With
your old setting you will occasionally get a slip. Normally not a major problem,
but it can wreck havoc with data transmissions (Unrestricted Digital) or in some
cases with the D channel signalling.

Peter



_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@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