[Asterisk-Users] Channelized T1, T100P problems

Pat Boyle pboyle at drizzle.com
Wed Apr 7 21:32:14 MST 2004


I've been having some problems getting a channelized T1 working with a T100P card. Perhaps someone can help:

I have an Eschelon T1 coming into a Vina Integrator box.  This box splits out the T1 into an ethernet plug for bandwidth and a secondary T1 which I plug into the T100P card. I've connected the two with a T1 crossover cable.  

I get a green light on the t100p card and can make outbound calls.  Inbound calls are not working as well.  Via the console, I can see them come in and start the "s" extension, but then they hang up.  Then the second channel accepts a call,  then I get a "all circuits busy" message on the phone (from my cell as I call in to test).  I'm testing from my cell phone and never hear a ring or any of the asterisk welcome message (even though it's showing in the console). It's as if the call appears to be answered by the card, but actually never is.

Changing the "immediate=yes" to "no" changes behavior somewhat.  Then I get a message in the console that there is no extension "6" defined, but if I define it, the problem doesn't go away.

I am using a channelized T1, not PRI.  So we don't have b and d channels.  Just e&m signalling.  Thanks for your help.
Pat Boyle

---------------------
System:
Eschelon T1, channelized (8 channels and the rest for bandwidth)
Dell Poweredge
2.4 Celeron
Redhat Fedora
Asterisk -0.7.2
Zaptel drivers 0.9.0

Here are the config files and console output:
-------------------------------

cat /etc/zaptel.conf
span=1,1,0,esf,b8zs
e&m=1-8
loadzone = us
defaultzone=us


cat /etc/asterisk/zapata.conf
[channels]
context=incoming
signalling=em_w
group=1
immediate=yes
channel => 1-8


extensions.conf
;the menu
[incoming]
exten => s,1,noop
; putting a wait here before would cause hangup issues if the person
; hung up right after the phone rang.  It seems that the card picks
; up (simple switch) but doesn't answer until the answer command.
; If you hang up before the answer command is executed, there are hangup issues.
exten => s,2,Answer
exten => s,3,DigitTimeout(5)    ; timout for dtmf entries, 5 is default
exten => s,4,ResponseTimeout(30); hangup after 30 sec of nothing
exten => s,5,Background(hello-recording);welcome... then wait for response or timeout


; press 1 for directory
; the argument should be the context specified in voicemail.conf
exten => 1,1,Directory(default)

; if invalid extention, then play invalid message and goto message again
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(s,5)

; on timout hangup
exten => t,1,Playback(goodbye)
exten => t,2,Hangup

exten => h,1,Hangup

 

-- from the console ---

*CLI>     -- Starting simple switch on 'Zap/1-1'
    -- Set Digit Timeout to 5
    -- Set Response Timeout to 30
    -- Playing 'hello-recording' (language 'en')
Urgent handler
    -- Starting simple switch on 'Zap/2-1'
    -- Set Digit Timeout to 5
    -- Set Response Timeout to 30
    -- Playing 'hello-recording' (language 'en')
Urgent handler
    -- Hungup 'Zap/1-1'
Urgent handler
Urgent handler
    -- Hungup 'Zap/2-1'
Urgent handler
Urgent handler
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040408/3c59e2ad/attachment.htm


More information about the asterisk-users mailing list