Configuring FreePBX for connecting to a traditional ISDN PBX

kalderista · July 9, 2011, 10:53pm

I’m configuring a freepbx server for managing incoming and outgoig calls fro isdn.

What I want is to receive calls trough one openvox B200E and send them to a traditional isdn PBX trough another openvox B200E card.

I have configured dahdi-channels.conf doing this:

First 2 spans as te mode
3th and 4th spans as nt mode, with “from-internal” cntext

Cards seem to be well configured, but i don’t know how to configure the web pannel to send all incoming calls from the firs card trough the secod isdn card.

Any hel please.

Thanks

SkykingOH · July 10, 2011, 5:15am

You need to put the outbound channels in their own group and build a trunk to that group.

Then use the inbound route module to send calls to the trunk.

This requires v2.8 or later.

kalderista · July 10, 2011, 9:13am

WOW!!

I did’nt realized this options exists!! very easy, fantastic!!

Thanks a lot!

kalderista · July 11, 2011, 3:56pm

Well,…

Inbound calls work fine, but I can’t redirect to the second B200E ISDN card

This is what I’ve done:

1.- Do dahid_genconf and edit dahdi-channels.conf. Span 3 and 4 ar in NT mode, so Span 1 and 2 are group 0 and span 3 and 4 are configured in group 1. Have alook at signalling and context. from-internal for span 3 and 4 is ok, isn’t it?:

; Span 1: B4/0/1 “B4XXP (PCI) Card 0 Span 1” (MASTER) RED
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel => 1-2
context = default
group = 63

; Span 2: B4/0/2 “B4XXP (PCI) Card 0 Span 2” RED
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel => 4-5
context = default
group = 63

; Span 3: B4/1/1 “B4XXP (PCI) Card 1 Span 1” RED
group=1,13
context=from-internal
switchtype = euroisdn
signalling = bri_net_ptmp
channel => 7-8
context = default
group = 63

; Span 4: B4/1/2 “B4XXP (PCI) Card 1 Span 2” RED
group=1,14
context=from-internal
switchtype = euroisdn
signalling = bri_net_ptmp
channel => 10-11
context = default
group = 63

2.- In freepbx web pannel, i’ve created a new trunk as ZAP Trunk (DAHDi compatibility Mode). Zap identifyer is g1. Do I’ve to do anything in Dialed Number Manipulation Rules? Do I’ve to configure DAHDI trunk instead of Zap Trunk?

3.- In inbound routes, I’ve configured one route named “ISDNINBOUND” and “set destination” >>>> Trunk > g1(zap)

If i make a call to freepbx machine I get this in the CLI:

– Executing [[email protected]:4] ExecIf(“DAHDI/i1/945044455-2b”, “1?Set(CALLERPRES()=allowed)”) in new stack
– Executing [[email protected]:5] Set(“DAHDI/i1/945044455-2b”, “DIAL_NUMBER=s”) in new stack
– Executing [[email protected]:6] GosubIf(“DAHDI/i1/945044455-2b”, “1?sub-flp-2,s,1”) in new stack
– Executing [[email protected]:1] ExecIf(“DAHDI/i1/945044455-2b”, “0?Return()”) in new stack
– Executing [[email protected]:2] Return(“DAHDI/i1/945044455-2b”, “”) in new stack
– Executing [[email protected]:7] Set(“DAHDI/i1/945044455-2b”, “OUTNUM=s”) in new stack
– Executing [[email protected]:8] Dial(“DAHDI/i1/945044455-2b”, “ZAP/g1/s,300,”) in new stack
[2011-07-11 16:19:38] WARNING[4223]: channel.c:5444 ast_request: No channel type registered for ‘ZAP’
[2011-07-11 16:19:38] WARNING[4223]: app_dial.c:2041 dial_exec_full: Unable to create channel of type ‘ZAP’ (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [[email protected]:9] Set(“DAHDI/i1/945044455-2b”, “CALLERID(number)=945044455”) in new stack
– Executing [[email protected]:10] Set(“DAHDI/i1/945044455-2b”, “CALLERID(name)=945044455”) in new stack
– Executing [[email protected]:11] Hangup(“DAHDI/i1/945044455-2b”, “”) in new stack
== Spawn extension (ext-trunk, tdial, 11) exited non-zero on ‘DAHDI/i1/945044455-2b’
– Hungup ‘DAHDI/i1/945044455-2b’

HELP PLEASE!! THANKS AGAIN!

SkykingOH · July 11, 2011, 4:48pm

You need to use a DAHDI trunk. ZAP has not been supported by Asterisk since 1.4.26

kalderista · July 11, 2011, 5:21pm

OK, but do I have to configure any rule??

SkykingOH · July 11, 2011, 5:26pm

You have to set your inbound route to terminate on the trunk.

If you have locally connected handsets you need to configure outbound route to point to trunk.

kalderista · July 11, 2011, 6:07pm

Still not working. It’s well configured as DAHDI trunk.

It seems that the dialplan is not well defined.

-- Executing [[email protected]:3] Goto("DAHDI/i1/945044455-5", "ext-trunk,tdial,1") in new stack
-- Goto (ext-trunk,tdial,1)
-- Executing [[email protected]:1] Set("DAHDI/i1/945044455-5", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [[email protected]:2] GotoIf("DAHDI/i1/945044455-5", "1?nomax") in new stack
-- Goto (ext-trunk,tdial,4)
-- Executing [[email protected]:4] ExecIf("DAHDI/i1/945044455-5", "1?Set(CALLERPRES()=allowed)") in new stack
-- Executing [[email protected]:5] Set("DAHDI/i1/945044455-5", "DIAL_NUMBER=s") in new stack
-- Executing [[email protected]:6] GosubIf("DAHDI/i1/945044455-5", "1?sub-flp-2,s,1") in new stack
-- Executing [[email protected]:1] ExecIf("DAHDI/i1/945044455-5", "0?Return()") in new stack
-- Executing [[email protected]:2] Return("DAHDI/i1/945044455-5", "") in new stack
-- Executing [[email protected]:7] Set("DAHDI/i1/945044455-5", "OUTNUM=s") in new stack
-- Executing [[email protected]:8] Dial("DAHDI/i1/945044455-5", "DAHDI/g1/s,300,") in new stack
-- Requested transfer capability: 0x10 - 3K1AUDIO

[2011-07-11 20:01:29] WARNING[6374]: sig_pri.c:6310 sig_pri_call: Unrecognized pridialplan NPI modifier: s
– Called g1/s
– Channel 0/1, span 3 got hangup, cause 18
– Hungup ‘DAHDI/i3/-2’

sniff…

kalderista · July 13, 2011, 2:02pm

Hi all,

I’ve been trying configuring chan_dahdi.conf using different values with pridialplan, but it doesn’t work

Does anyone know this isue?

I tried instaling a differente versión of asterisk (asterisk-es-rsp) but I don’t get any solution.

Is it a ISDN card configuration isue? Inboun calls configuration isue?

any heklp will be wellcome.

Thanks again.

SkykingOH · July 13, 2011, 6:33pm

Cause code 18 is no user responding.

Not knowing your old switch this is difficult to debug in a forum. You may need to dig into the PRI debug logs or possibly even take a look with an ISDN protocol analyzer. I can’t imagine trying to troubleshoot issue like this without the right tools are an in depth knowledge of the subject.

Understand why you are not getting much help. You are trying to get something that is by no means an off the shelf configuration to work. In order to accomplish this you need an Asterisk expert, a transport/signalling expert and an expert on the legacy system.

kalderista · July 13, 2011, 10:00pm

Do you mean it’s not asterisk PBX isue? The other PBX is an old 1,4 aserisk pbx with a sempron procesor.

This old aserisk PBX is working fine now, in ptmp mode, and I’m trying to configure a this new freepbx server for a real old ISDN PBX for build a hybrid Voip system.

Should it work with the PBX?

I will try to build a new asterisk servers to build this scenary. We will see…

Thank you anyway.

kalderista · September 13, 2011, 9:59pm

Finaly I got it work, creating a custom context in extensions_custom.conf like this:

[TO-OLD-PBX]

exten => _X.,1,NoOp()
exten => _X.,n,Dial(DAHDI/g1/${EXTEN},45,tTwW)
exten => _X.,n,Hangup()

Sending a trunk for sending calls to another trunk does definitivly not work!

system · June 4, 2014, 7:17pm