DAHDI Dialing Out Problem


Fresh install:
Asterisk SVN-branch-1.6.1-r211958
DAHDI compatibility mode (ZAP2DAHDICOMPAT=true)

DAHDI incoming works fine, can’t dial out. Get these warnings:
channel.c: No channel type registered for 'ZAP’
app_dial.c: Unable to create channel of type ‘ZAP’ (cause 66 - Channel not implemented)

This problem caused by the following entry in extensions_additional.conf
OUT_1 = ZAP/g0

Changing ZAP/g0 to DAHDI/g0 fixed the problem but the file (extensions_additional.conf) is overwritten when updating the configuration.

Any help is highly appreciated.

Solved by:
echo “OUT_1 = DAHDI/g0” >> /etc/asterisk/globals_custom.conf

Anyone has a better solution please let me know

They broke it! I caught a lot of crap today 'cause I didn’t notice that when I updated Freepbx over the weekend that outbound calls no longer worked! (talk about getting a nice slap in the face!)

After digging through the code on our backup (which we had to switch back to) you have to add 3 lines back into admin/modules/core/functions.inc.php

In the latest version of freepbx, they removed these lines. Starting at 1234.

The older version of the file has this line at 1172

Here is how the whole section should look.

foreach($globals as $global) { $value = $global['value']; if ($chan_dahdi && substr($value, 0, 4) === 'ZAP/') { $value = 'DAHDI/' . substr($value, 4); } $ext->addGlobal($global['variable'],$value);

The newer version is missing the three middle lines and looks like this

foreach($globals as $global) { $value = $global['value']; $ext->addGlobal($global['variable'],$value);

Put that back in and it properly creates the OUT_1 = DAHDI/g0 lines.

This was fixed within about 10 minutes of being reported, you should be able to update to get the fix, have a look online.


oops, it was actually 14 minutes form when it was reported :slight_smile:

DID just stands for direct inward dial. It refers to a number associated to a trunking facility. It’s not an Asterisk or VoIP term.

DID’s are differentiated based on the Dialed Number identifiaction digits in the trunk (this can be inband or via a PRI ISUP message).

don’t know if im hijacking this threat, but im having this problem i know that this is an old threat i was looking for more info but could not find it, i have a T1 using Dahdi, everytime when i sumit a change in freepbx the extension_additionals.conf get overwrite, I have to edit the file manually and change the OUT_4 = ZAP/g0 to OUT_4 = DAHDI/g0 to works,

how can i fix this?

Here is the info:

Asterisk 1.4.42


Go to the advanced settings module and turn DAHDI mode on.

You did not mention which version of FreePBX. You need to set the system in DAHDI mode. You have it in Zap compat.

I don’t see the option under FreePBX Advanced Settings.

the only option relate it to DAHDI are this 2

SIP and DAHDi callgroup
SIP and DAHDi pickupgroup

sorry bro i’m relative new on DAHDI hardware setup.

Thanks again

It’s under “dialplan and operational”, third setting down “convert zap to DAHDI”

At the top of the advanced configuration page you have to make the advanced options visible.

I am new hunk to the PBX world,i hv setup successfuly sip communication over the internet till now.Now i am looking forward to route calls on PSTN for outbound and inbound.CONFUSION is…
1-after adding and configuring hardware like TDM400p and sangoma A102D to my PBX,would i be able to make and recieve calls from PSTN or i must need to buy DIDs for this purpose??

you will need lines from the PSTN companies like verizon, depending on the service like PRI/T1 if it is analog lines then the lines comes with DID already

you are the best…

ok wait a second,what if i put my telephone line to FXO port into the card,would i be able to call or still i need to contact my telephone company to provide DIDs?

and also received calls to the number that comes with your analog line(the one that you will connect to the FXO Port)

you just have to create a Trunk for the Card and an outbound and inbound route

But i still have half of the confusion there about DID?please give a little clear description of DID,and why we need that e.g in my scenario?i hav studied stuff about DID,but i want description from asterisk or voip guy… Thnx in advance

Ok,thnx for the previous help,m goin on the right track till now.I am goin to configur sangoma A200D for my PBX,ONE THING I WANT TO KNOW …when calls will be routed on my Phone line,would there be any charges which will be included in my phone bill at the end of the month?its because of i am using that line to make calls outward…thnx in advance

From your phone companies perspective a call is a call. They don’t know if it is a PBX or a $2 walmart phone.