Mehran Mozaffari
2005-Jul-05 04:43 UTC
[Asterisk-Users] Newbie question reg. Asterisk and Channel Access Bank I and TE110p
Hi,
I have some problem to get this setup working. I have a CAC Channel
Banl I, with FXO and an Asterisk box ( I am using Asterisk@Home 1.2)
and I have a TE110p installed in this box.
What I want to do is, just to be able to dial one of those lines that
already are connected to the channel bank, and transfer that call
through TE110p and Asterisk to a user agent somewhere through
Internet.
<PSTN>-----<CAC Channel Bank
I>-----<TE110p>-----<Asterisk>-----<SIP User Agent>
At this time the SIP UAs can communicate with each other and
everything works properly, but I can't dial through channel bank. When
I dial one of those numbers, I will get no answer ring, and I can't
see anything coming to Asterisk through CLI. and when I tried to dial
through SIP UA to the PSTN end, I will get all circuits are busy now
from asterisk.
Any Idea what should I do? at this time all lights are green and it
looks like that everything is working properly, but I am not sure
where is the problem, here are my settings:
/etc/zaptel.conf:
----------------------------------------
span=1,1,0,esf,b8zs
fxols=1-24
/etc/asterisk/zapata.conf:
----------------------------------------
[channels]
signalling=fxo_ls
context=default
group=1
channel = 1-24
On the CAC, I set the rear (side without plugs) panel switches to all
off and the front panel slide switches to all normal. The front panel
dip switches has set to 0111100000 (1234567890).
This is what I got in console:
----------------------------------------
-----------------------
-- Executing Macro("SIP/200-1661",
"dialout-trunk|1|8058111|") in new stack
-- Executing GotoIf("SIP/200-1661", "1?3:2)") in new
stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/200-1661",
"record-enable|200|OUT") in new stack
-- Executing GotoIf("SIP/200-1661", "0 > 0?2:4") in
new stack
-- Goto (macro-record-enable,s,4)
-- Executing GotoIf("SIP/200-1661", "1?5:8") in new
stack
-- Goto (macro-record-enable,s,5)
-- Executing DBget("SIP/200-1661",
"RecEnable=RECORD-OUT/200") in new stack
-- DBget: varname=RecEnable, family=RECORD-OUT, key=200
-- DBget: Value not found in database.
-- Executing SetVar("SIP/200-1661",
"CALLFILENAME=OUT200-20050704-043103-1120465863.0") in new stack
-- Executing Goto("SIP/200-1661", "s|14") in new stack
-- Goto (macro-record-nable,s,14)
-- Executing GotoIf("SIP/200-1661", "0?15:99") in new
stack
-- Goto (macro-record-enable,s,99)
-- Executing NoOp("SIP/200-1661", "NO RECORDING NEEDED")
in new stack
-- Executing GotoIf("SIP/200-1661", "1?7") in new stack
-- Goto (macro-dialout-trunk,s,7)
-- Executing GotoIf("SIP/200-1661", "1?9") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing SetGroup("SIP/200-1661", "OUT_1") in new
stack
-- Executing CheckGroup("SIP/200-1661", "3") in new
stack
-- Executing SetVar("SIP/200-1661",
"DIAL_NUMBER=8058111") in new stack
-- Executing SetVar("SIP/200-1661", "DIAL_TRUNK=1") in
new stack
-- Executing AGI("SIP/200-1661", "fixlocalprefix") in
new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("SIP/200-1661", "OUTNUM=8058111") in
new stack
-- Executing Cut("SIP/200-1661", "custom=OUT_1|:|1") in
new stack
-- Executing GotoIf("SIP/200-1661", "0?19") in new stack
-- Executing Dial("SIP/200-1661", "ZAP/g0/8058111") in
new stack
== Everyone is busy/congested at this time
-- Executing Goto("SIP/200-1661", "s-CHANUNAVAIL|1") in
new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/200-1661", "Dial failed due to
CHANUNAVAIL") in new stack
-- Executing Macro("SIP/200-1661", "outisbusy") in new
stack
-- Executing Playback("SIP/200-1661",
"allison7/all-circuits-busy-now") in new stack
-- Playing 'allison7/all-circuits-busy-now' (language 'en')
-- Executing Playback("SIP/200-1661",
"allison7/pls-try-call-later") in new stack
-- Playing 'allison7/pls-try-call-later' (language 'en')
-- Executing Macro("SIP/200-1661", "hangupcall") in new
stack
-- Executing ResetCDR("SIP/200-1661", "w") in new stack
-- Executing NoCDR("SIP/200-1661", "") in new stack
-- Executing Wait("SIP/200-1661", "5") in new stack
-- Executing Hangup("SIP/200-1661", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/200-1661' in macro
'hangupcall'
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on
'SIP/200-1661' in macro 'outisbusy'
== Spawn extension (from-internal, 98058111, 2) exited non-zero on
'SIP/200-1661'
-- Executing Macro("SIP/200-
1661", "hangupcall") in new stack
-- Executing ResetCDR("SIP/200-1661", "w") in new stack
-- Executing NoCDR("SIP/200-1661", "") in new stack
-- Executing Wait("SIP/200-1661", "5") in new stack
-- Executing Hangup("SIP/200-1661", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/200-1661' in macro
'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/200-1661'
asterisk1*CLI>
----------------------------------------
Sorry for long email,
I would appreciate if somebody that use these devices can help me in
right direction.
Regards,
Narhem
Julian J. M.
2005-Jul-05 05:25 UTC
[Asterisk-Users] Newbie question reg. Asterisk and Channel Access Bank I and TE110p
Recheck your zaptel.conf. That's not the correct setup for a T1 trunk. You need to know the signalling the channel bank uses, and specify the voice channels (bchannel=1-24), and the signalling channel (dchannel=25). Those numbers are bogus, as I've never worked with T1 ;) BTW, why are you using such setup (1 channel bank to connect to 24 analog lines) instead of asking your Telco to install a T1 trunk in your office? Julian. On 7/5/05, Mehran Mozaffari <mehran.mozaffari@gmail.com> wrote:> Hi, > > I have some problem to get this setup working. I have a CAC Channel > Banl I, with FXO and an Asterisk box ( I am using Asterisk@Home 1.2) > and I have a TE110p installed in this box. > > What I want to do is, just to be able to dial one of those lines that > already are connected to the channel bank, and transfer that call > through TE110p and Asterisk to a user agent somewhere through > Internet. > > <PSTN>-----<CAC Channel Bank I>-----<TE110p>-----<Asterisk>-----<SIP User Agent> > > At this time the SIP UAs can communicate with each other and > everything works properly, but I can't dial through channel bank. When > I dial one of those numbers, I will get no answer ring, and I can't > see anything coming to Asterisk through CLI. and when I tried to dial > through SIP UA to the PSTN end, I will get all circuits are busy now > from asterisk. > > Any Idea what should I do? at this time all lights are green and it > looks like that everything is working properly, but I am not sure > where is the problem, here are my settings: > > /etc/zaptel.conf: > ---------------------------------------- > span=1,1,0,esf,b8zs > fxols=1-24