Question : d channnel is down solved

thank you very much, my friend RichardHH
I will try it as you said. Pl continue to pay attention to this post, I ll report my progress and problems

hi RichardHH
Now I see the NTBA box is used to support the power for the isdn phones. So I guess when I use the * box as a gateway, which sit before the traditional PBX (as I said before), then the additional NTBA is not necessary anymore, because the traditional PBX has already it’s own power , and then things will be easier, that means I can just simple plug the isdn cable of the old PBX direct into the HFC card, as if it was a isdn telephone. is this thinking right?
What I m now wondering is why I can not call from outside through the isdn cabel into the internal sip hard phones . When I called , I just heared the busy tone

And I m still not clear about the cross-over-cable, what is the functionality of the cross-over-cable?

I m trying to call from outside through asterisk to internal sip phones. I thinks the .conf files should be ok, because some calls really reached the sip phones, the problem is that the d channel is not stable.
here is the story of my test:
I have two isdn cables , and used one of it first to plug into TE port 1, the d -channel is down , and i don’t know how to bring it up. so I tried it on the second TE port, this time the d channel is on, and i can call from outside into internal.

and then I tried the second cable while the first still there, but the second cable can not bring the d channel up , (the d channel of cable 1 is still up).

then I tried remove all the two cables, and plug them again, this time ,both d - channel are down

the second question is, I can not call from inside to outside with sip phone(i tried while one d channel was still up)
extensions.conf:

[outgoing]
exten => _X.,1,Dial(Zap/g1/${EXTEN},30)
exten => _X.,2,Hangup()

[internal]
include => outgoing

where is the problem?

[quote=“bxjy”]hi RichardHH
Now I see the NTBA box is used to support the power for the isdn phones. So I guess when I use the * box as a gateway, which sit before the traditional PBX (as I said before), then the additional NTBA is not necessary anymore, because the traditional PBX has already it’s own power , and then things will be easier, that means I can just simple plug the isdn cable of the old PBX direct into the HFC card, as if it was a isdn telephone. is this thinking right?
[/quote]

Not nessecarily: It depends on the PBX. Which PBX is it ?
We need to know what PBX it is and what interface.
Is it something like an Eumex with standard “comfort ISDN Anschluss” or a siemsn hipath with an SM2 interface ? The cabeling is already different.

Then we need to setup the PBX and asterisk, that asterisk allows call FROM the pbx and the pbx allows incomings (MSN config) FROM Asterisk.

For that, we need to see the CLI output too.

But first, we need to know what PBX it is exactly.

the PBX we are using is “Telefonieserver Alcatel Omni-PCX”

and now the PBX is still not connected with *box, i m trying to use the sip phone through a seperate isdn cable to call out.
I hope the name of the PBX give some senses to figure out what kind of cable will be used.

Alcatel…ok.

Honestly, may i ask why you want to keep using it ?
WOuldnt it be better/cleaner to make an Asterisk setup and using SIP hardphones ?

If not, i look up the Alcatel and its interface - so: Do you HAVE to keep going with the alcatel ?

PS:

Is it the 4200 or 4400 ?

NM…no matter if 4200 or 4400, i just figure another problem:

If you want to keep the alcatel-pbx, you will face another problem:
The lines…

See, the alcatel-pbx is using interface moduls for the lines incoming/outgoing.
So the capacity of call lines is NOT determined by asterisk, its determined by the alcatel-pbx.

Imagine you have 1 modul in the alcatel-pbx: Then only (max) 2 phones can talk simultan via the “bridge” to asterisk.

And if you need 8 lines (4 modules), you need the same capacity card in the asterisk box.
And honestly, thats crap !

I would REALLY suggest, building a native Asterisk/SIP pbx environment and place the alcatel at ebay…

it’s not up to me to use or not use Alcatel, so what I must do is to handle the problem.
I m not clear about the Interface of Alcatel, but I will ask and paste here a little bit later

I have already a 4-ports HFC Junghanns isdn card, and the Alcatel has really 4 incoming ISDN cable from telco(as you said 4 moduls). But why is this combination a crap?
I m wondering , because I thought the *box is good at this kind of combination. Can you pl explain it?

go on pasting another question,why the sip phone wont go out?
in sip.conf the sip phone has the context : internal

in [internal] I set :include => outgoing

in [outgoing] I set:exten => _X.,1,Dial(Zap/g1/${EXTEN},30)

in zapata.conf I set:

now when I use sip phone to call out, i see on the screen of *box that it is trying to execute zap/g2/number, even I comment out the g2

I have looked up the “alcatel omni-pcx” on the web of alcatel, and found it is neither 4200 nor 4400, it is another type. I m not clear if it is a omni-pcx enterprise or omni-pcx office, dose this information help?

here is the output of *box ,after trying to call outside

what dose this mean?

ok, now the outgoing calls from sip phones work now.
it was only a typo .

now I m trying to combine *box with the alcatel box
I plug a isdn cable from ntba into TE port , and a incoming cable of alcatel in NT port
when I called, it ring once then turned busy tone.
anybody knows why?

output of trying to call from outside into alcatel

[quote=“bxjy”]here is the output of *box ,after trying to call outside

what dose this mean?[/quote]

That there is no ZAP interface asterisk could access.
Means the ISDN line isnt up !

[quote=“bxjy”]ok, now the outgoing calls from sip phones work now.
it was only a typo .[/quote]

Oh, ok , yay :smile:

[quote=“bxjy”]output of trying to call from outside into alcatel

[quote]
– Accepting voice call from ‘3714008481’ to ‘400090’ on channel 0/1, span 2 – Executing Answer(“Zap/4-1”, “”) in new stack
– Executing Dial(“Zap/4-1”, “Zap/g2/0371400090|20|r”) in new stack
– Requested transfer capability: 0x10 - 3K1AUDIO
– Called g2/0371400090
– Channel 0/1, span 3 got hangup, cause 42
– Zap/7-1 is circuit-busy
– Hungup ‘Zap/7-1’
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel ‘Zap/4-1’ status is ‘CONGESTION’
– Channel 0/1, span 2 got hangup request
– Hungup ‘Zap/4-1’

[/quote][/quote]

Cause 42 means layer 3 error. Mostly=No B-Channel
Ok, lemme see what we do here…

Hmm…what i dont get:
Accepting ->VOICECALL<- on…(CLI output), then its switching to bearer 3K1 braring (Fax)…was that a fax you made the test with ?

Ok …CLI output line for line, im confused and THIN, the order of the interfaces (groups) in the zaptel.conf is wrong (just swap block 1 and block 2).

Ok…Line for line:

– Accepting voice call from ‘3714008481’ to ‘400090’ on channel 0/1, span
2

Okies, there is the leading 0 missing but thats cuz u are obv. using TRUNKMSD=1 in your dialplan, so NP here.

Ok…means, a call is coming in on Span 2 (what is span 2 in your zapata?) channel 1.

– Executing Answer(“Zap/4-1”, “”) in new stack
Here begins my confusion, it should read
– Executing Answer(“Zap/2-1”, “”) in new stack
Where does the “4” come from ?!

– Executing Dial(“Zap/4-1”, “Zap/g2/0371400090|20|r”) in new stack
Completely baffled now…
When span 2 is (see above) the card for the incoming call which should go to a SIP phone, why do we dial out on ZAP/g2 (which is span2) again ?!

Ok, fullstop captain, can your post your !!CURRENT!! extension.conf, zaptel.conf and zapata.conf please :smiling_imp:

You get me confused now…