I'm Running elastix 2.2.0 14, i have no idea what to do to get my dahdi line to ring my extention....

What do i do?

What do i do?

Provide more information..

What sort of Dahdi line (PSTN/POTS , E1, T1)?
Do currently have calls coming in on the Dahdi line to IVR or Ring Groups??
Are you trying to get a particular DID to ring an extension directly??

Otherwise, to get a dahdi line to ring an extension, you make sure you have Dahdi Trunk setup, setup an inbound route and point that inbound route to a extension destination.

I am assuming that you have a new setup, so you may need to setup an catchall inbound route which means that you set an inbound route with no DID and no CID set, which will capture all calls coming into your Elastix server.

Anyhow more information would be good....

Regards

Bob

I have a asterix card (4port) with 2 single channel cards, i eventually got them to all forward to the same extention.... but i need them to be seperate as in....

DAHDI 1 to go to extention 1000
Dahdi 2 to go to extention 8999
sip account 1 to go to extention 2000

all i got working was the "all" one....

so all incomming calls is goint to a switchboard 1000 at the moment

i'm still learning elastix / Free pbx so please be patient

i have international static I.P's available but really dont want to publish it here....

Thank you

bertuslouw,

An Asterisk 4 port card can also mean a 4 port E1/T1 card, but I am going to assume that you mean a 4 Port PSTN card like a TDM410 or equivalent based on the rest of your email.

What you are trying achieve is possibly with PSTN, but you have to perform some manual mapping of the PSTN lines to DID numbers, as PSTN (generally) does not provide the DID numbers that the caller called.

At the moment, when you setup a Hardware Telephony card, automatically there is a Dahdi/G0 setup, which is basically for all the lines on that card (G = group).

The first step is to setup individual dahdi lines for that card. So going in Trunk setup and make a Dahdi Trunk, give it a name e.g. dahdi1, and in the Dahdi identifier just put the number 1, like wise with the next line e.g. dahdi2, and in the identifier put the number 2. Please note here that I assuming that the ports you are using for the two lines are port 1 and port 2, if you have them in 3 and 4, naturally you need to change the numbers above.

Now you have setup separate dahdi trunk lines...

Thie next step is optional, especially if all your outbound calls are going out via the SIP Trunks. But if you want to use your PSTN lines for outbound, in your outbound routes, instead of just referencing Dahdi/G0 as your trunk for outgoing, you would now have e.g.

Trunk 1 : dahdi1
Trunk 2 : dahdi2

Right, we are finished with the optional, I have just mentioned it for anyone else reading this post...

The critical step which you need to implement so that your system can determine which line the call come in on is by going into the PBX menu and select ZAP Channel DIDs

You will see a some entries. To setup line 1 (port 1), you would enter

Channel : 1 {this correlates to the dahdi number you setup in the trunks}
Description : Johns Line {whatever description you want)
DID : 0244922993 {this does not have to be the line number it is just number}

and do the same for Channel : 2

Channel : 2 {this correlates to the dahdi number you setup in the trunks}
Description : Johns Line {whatever description you want)
DID : 0244922999 {this does not have to be the line number it is just number}

Now I mentioned that the DID does not have to match the line phone number, as it is just a reference number, but why complicate things, and makes it easier sorting out issues, so use the exact phone number of the line that is in port 1.

Now in the inbound routes, you can now enter a DID (which are the DID's you setup in ZAP Channel DIDs).

So in the inbound routes, you setup DID : 0244922993 / CID {leave empty} and at the end destination 1000

and the same again for the other DID you setup DID : 0244922999 / CID {leave empty} and at the end destination 8999

Now one final thing, you need to manually edit a file called

/etc/asterisk/dahdi-channels.conf

in under each port, you will see that the context=from-pstn, this needs to be changed to context=from-dahdi (context used to be from-zaptel and this still works as well).

Once you have made this change, then reboot, and try it...

Finally, you are wondering about the SIP channel, how to direct that to extension 2000.
Well SIP is a digital line, just like E1/T1 which present their DID when the call comes in. So have a look in your /var/log/asterisk/full log after you have made a call into the Elastix system on the SIP line, and you should see that it presented a DID number (usually saying that you should have an entry for this). Add this DID into your inbound routes, with a destination of 2000 and that should work fine, alternatively, if this is you only other line, you can leave it working on the catch all , and point the catchall to extension 2000....

Hope this all makes some sort of sense, if not post back.....

Regards

Bob

Bob,
how about Zap trunk (DAHDI Compatibility mode), default is Zap/g0, I can do it the same with DAHDI trunk? Zap trunk can receive calls and pass voices, but CID not display...Any suggestion for this?

dungth wrote:
Bob,
how about Zap trunk (DAHDI Compatibility mode), default is Zap/g0, I can do it the same with DAHDI trunk? Zap trunk can receive calls and pass voices, but CID not display...Any suggestion for this?

Yes, I recently solved that, you need to enter in Elastix GUI to ZAP channel (identify first what channel you want to use), Create a new channel, put the channel and the external number of the pstn, and then make an inbound route with the pstn number and redirect it to the extension.

At last you need to modify dahdi-channels.conf find the channel and in context put from-zaptel.

Regards
Camus Red.

----------------------------------------
the following of configuration steps:
----------------------------------------

step1: creat 2 dahdi trunk

trunk name: dahdi 13 dhadi identifier: 13
trunk name: dahdi 14 dhadi identifier: 14
all on trunk CID is empty blank

step2: creat zap channels DID

channel: 13
description: channel 13
DID: 848013 (not line number, it just is an any number)

channel: 14
description: channel 14
DID: 848014 (not line number, it just is an any number)

step3: creat inbound route

setup DID : 848013 / CID {leave empty} and at the end destination 100 (softphone to register extension-SIP)

setup DID : 848014 / CID {leave empty} and at the end destination 101 (softphone to register extension-SIP)

step4: edit dahdi-channel.conf
change context=from-pstn to be context=from-zaptel

OK. reboot..yeah

I make a call from inside Ext 100 to outside -> OK.
But a call from outside to inside (Ext 100) -> notice busy, no answer and no CID.

--------------------------------------------
Code: a call from outside into inside Ext 100
--------------------------------------------

-- Starting simple switch on 'DAHDI/13-1'
-- Executing [s@from-zaptel:1] NoOp("DAHDI/13-1", "Entering from-dahdi with DID == ") in new stack
-- Executing [s@from-zaptel:2] Ringing("DAHDI/13-1", "") in new stack
-- Executing [s@from-zaptel:3] Set("DAHDI/13-1", "DID=s") in new stack
-- Executing [s@from-zaptel:4] NoOp("DAHDI/13-1", "DID is now s") in new stack
-- Executing [s@from-zaptel:5] GotoIf("DAHDI/13-1", "1?dahdiok:checkzap") in new stack
-- Goto (from-zaptel,s,9)
-- Executing [s@from-zaptel:9] NoOp("DAHDI/13-1", "Is a DAHDI Channel") in new stack
-- Executing [s@from-zaptel:10] Set("DAHDI/13-1", "CHAN=13-1") in new stack
-- Executing [s@from-zaptel:11] Set("DAHDI/13-1", "CHAN=13") in new stack
-- Executing [s@from-zaptel:12] Macro("DAHDI/13-1", "from-dahdi-13,s,1") in new stack
-- Executing [s@macro-from-dahdi-13:1] NoOp("DAHDI/13-1", "Entering macro-from-dahdi-13 with DID = s and setting to: 848013") in new stack
-- Executing [s@macro-from-dahdi-13:2] Set("DAHDI/13-1", "__FROM_DID=848013") in new stack
-- Executing [s@macro-from-dahdi-13:3] Goto("DAHDI/13-1", "from-trunk,848013,1") in new stack
-----------------------------------------------------
-- Goto (from-trunk,848013,1)
== Channel 'DAHDI/13-1' jumping out of macro 'from-dahdi-13'
-- Executing [848013@from-trunk:1] Set("DAHDI/13-1", "__FROM_DID=848013") in new stack
-- Executing [848013@from-trunk:2] Gosub("DAHDI/13-1", "app-blacklist-check,s,1") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("DAHDI/13-1", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("DAHDI/13-1", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("DAHDI/13-1", "") in new stack
-- Executing [848013@from-trunk:3] ExecIf("DAHDI/13-1", "1 ?Set(CALLERID(name)=)") in new stack <--- no CID from Telco
-- Executing [848013@from-trunk:4] Set("DAHDI/13-1", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [848013@from-trunk:5] Set("DAHDI/13-1", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [848013@from-trunk:6] Goto("DAHDI/13-1", "from-did-direct,100,1") in new stack
-------------------------------------------------------
-- Goto (from-did-direct,100,1)
-- Executing [100@from-did-direct:1] Macro("DAHDI/13-1", "exten-vm,novm,100") in new stack
-- Executing [s@macro-exten-vm:1] Macro("DAHDI/13-1", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("DAHDI/13-1", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("DAHDI/13-1", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("DAHDI/13-1", "1?Set(REALCALLERIDNUM=)") in new stack
-- Executing [s@macro-user-callerid:4] Set("DAHDI/13-1", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("DAHDI/13-1", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("DAHDI/13-1", "1?report") in new stack
........
........
-- Goto (macro-dial-one,s,41)
-- Executing [s@macro-dial-one:41] ExecIf("DAHDI/13-1", "1?Set(DIALSTATUS=NOANSWER)") in new stack <--- 3CX Ext 100 does not receive call
-- Executing [s@macro-dial-one:42] NoOp("DAHDI/13-1", "Returned from dial-one with nothing to call and DIALSTATUS: NOANSWER") in new stack
-- Executing [s@macro-dial-one:43] MacroExit("DAHDI/13-1", "") in new stack
-- Executing [s@macro-exten-vm:10] GotoIf("DAHDI/13-1", "0?exit") in new stack
-- Executing [s@macro-exten-vm:11] Set("DAHDI/13-1", "SV_DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("DAHDI/13-1", "0?docfu,1") in new stack
-- Executing [s@macro-exten-vm:13] GosubIf("DAHDI/13-1", "0?docfb,1") in new stack
-- Executing [s@macro-exten-vm:14] Set("DAHDI/13-1", "DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:15] NoOp("DAHDI/13-1", "Voicemail is 'novm'") in new stack
-- Executing [s@macro-exten-vm:16] GotoIf("DAHDI/13-1", "1?s-NOANSWER,1") in new stack

???

Greetings dungth,

It worked this method for me when I used the line number in Zap Channel DIDs and in the inbound route directed to the extension. I got confused in your previous post. Did you use Zapata channels with Dadhdi compability?

You could try to use your line number for testing purposes.

Regards,
Camus Red.

Dear

I have same issue, i cant make a call from.

I did same as mention.

I am using Synway FXM3200P card with PAP2T ata.

Can you suggest how to configure ????

Please suggest me, how to connect ata to elastix.

Please help me.

Regards,