DAHDI Dialing Out Problem

Hi,
Fresh install:
Asterisk SVN-branch-1.6.1-r211958
FreePBX 2.5.1.5
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
[globals]
OUT_1 = ZAP/g0Changing 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
Solved by:
echo "OUT_1 = DAHDI/g0" >> /etc/asterisk/globals_custom.confAnyone has a better solution please let me know
Thanks
Better solution...
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
This was fixed within about 10 minutes of being reported, you should be able to update to get the fix, have a look online.
UPDATE:
oops, it was actually 14 minutes form when it was reported :)
Philippe Lindheimer - FreePBX Project Leader
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Hey Philippe
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?
You did not mention which
You did not mention which version of FreePBX. You need to set the system in DAHDI mode. You have it in Zap compat.
upss sorry forgot
Here is the info:
FreePBX 2.9.0.7
Asterisk 1.4.42
Thanks
Go to the advanced settings
Go to the advanced settings module and turn DAHDI mode on.
Hey SkyKing thanks in advance for your help
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
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.
Thanks it workss...
you are the best..
Confusion about DID(direct inward dialing)
Guys,
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??
if you install a card
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
DIDs come with my telephone line?
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?
Yes you will be able to call
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
Really...I am glad to read that,Thnx man...
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
DID just stands for direct
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).
My Phone line is used for asterisk
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
From your phone companies perspective a call is a call. They don't know if it is a PBX or a $2 walmart phone.