Hi guys. I’m quite new in asterisk. pretty much a newbie and im having problem setting up asterisk.
I’ve successfully installed AsteriskNow 1.6 and FreePBX which it’s currently running now and accessible through web browser on the other pc.
I have a TDM400p Analog hardware device which contains 2 each of fxo & fxs port and is connected on my computer where asterisk is currently installed. The system detects my analog device on the Dahdi.
But I’m really getting confused on where to start. How will I get the device work. where I will start? what should I do first? what else i’m missing?.
Any reply is very much appreciated. thanks in advance.
Since you have FXS ports, hook your traditional analog phones to them and set them up in the web interface and get them working between each other.
Get some SIP phones working, both hard and soft.
Connect the FXO port(s) to line(s) coming in the the PSTN. Define the inbound routes and outbound routes to handle call flow.
These are the basic steps. I leave the specifics of how to do each step to you. There is lots of information out there on doing so. Also, though it does not detail working with AsteriskNow or FredPBX, I recommend reading the Asterisk book.
I understand that in in-order to get it work, I had to configure the trunk, extension, inbound/outbound routes.
But the trunk type confuses me, since I will use my fxs and fxo port (tdm400p). What trunk will I use when adding new, Is it categorized on DAHDI or ZAP?
Though you can probably add the trunk as a ZAP with Dahdi compatibility mode, it would probably be best to add it as a DAHDI trunk as DAHDI is the current driver and is most likely what was installed.
FYI. Basically, both ZAP and DAHDI are drivers for the same cards. The drivers were originally the ‘ZAPtel’ drivers. There were some issues relating to the use of the trem ZAPtel, so the drivers were renamed to DAHDI. Asterisk has different commands and device names based on which version is in use, ZAP or DAHDI. FreePBX can work with either, but if you use the ZAP trunk while using the DAHDI drivers, the dial plan code converts one name to the other. That is my understanding of the compatibility mode any why.
[quote=“dalenoll”]Though you can probably add the trunk as a ZAP with Dahdi compatibility mode, it would probably be best to add it as a DAHDI trunk as DAHDI is the current driver and is most likely what was installed.
FYI. Basically, both ZAP and DAHDI are drivers for the same cards. The drivers were originally the ‘ZAPtel’ drivers. There were some issues relating to the use of the trem ZAPtel, so the drivers were renamed to DAHDI. Asterisk has different commands and device names based on which version is in use, ZAP or DAHDI. FreePBX can work with either, but if you use the ZAP trunk while using the DAHDI drivers, the dial plan code converts one name to the other. That is my understanding of the compatibility mode any why.[/quote]
Alright tfti. uhm is it possible to use my fxo port(configured in my trunk/dahdi) as the line that will be used by my softphones configured extension for their outbound calls?
Yes. You do that by setting up an outbound route. Define the dialling pattern that needs to be matched and select your DAHDI trunk(s) under ‘trunk sequence’. If you have more than one trunk, specify them in the order you want them used. If the first is busy, then the second will be attempted.
If you unfamiliar with dialling pattern, hover your mouse over the section heading in the outbound route definition screen and it will pop up a help screen. For example, to match a 7 digit US number, the pattern would be NXXXXXX. To match standard US long distance (including the country code), or a 10 digit dial, the pattern is 1NXXNXXXXXX.
I’ve already plugged my phone line into my fxo port(tdm400p, channel 1) that is configured and define in my trunk/zap in channel 1 also. Is there a way to test that it is working?
Already tested my phone line which is plugged in fxo port of my tdm400p in analog phones and it works fine.
Iv’e tried configuring an outbound route that points to that trunk. with a dialing pattern of:
Does this means that if I dialed 9 + the 7 digit number of phone. it matches the route?
btw I’m not in the US.
I tried to call from my softphone(x-lite) to another softphone extension within the network works fine. But when I tried to make an outside call using that pattern say’s “all circuits are busy now please try again later.” but I’m pretty much sure the number I’m dialing is not in use.
[quote=“dalenoll”]The basic tasks that I would recommend would be.
Since you have FXS ports, hook your traditional analog phones to them and set them up in the web interface and get them working between each other.
[/quote]
BTW I noticed when I hooked my two traditional analog lines to the fxs ports of my tdm400p, there is no dial tone or any sound can be heard on the handset.
There is already an Extension/zap configured which points to the two fxs ports of tdm400p(port 3 and 4).
Is there anything I should do to get it work?
Dont mind this post. Already figured out the problem. Thanks
If you are connecting your FXO port directly to the PSTN, remove the prepend and leave it blank. What you have does match 9 + 7 digits, but that whay you have it written, the system will strip off the 9 that you dialed(prefix) and replace it with a 9(prepend).
All circuits busy indicates that there are no trunks available for that route, not that the far end is busy. If you defined your trunk(s) and have the route set to use those trunk(s), then perhaps the problem is at a lower level such as the configuration of the card (/etc/dahdi/system.confg, /etc/dahdi/modules.conf, /etc/asterisk/chan_dahdi.conf or /etc/asterisk/dahdi-channels.conf).
This will trigger the route If I dial the 7 digit number?
I recently updated my dahdi module to ver 2.8 does it affect the configuration? how do I check if the configs are correct?
and lastly. What are the things that needs to be configured in order to make an outside call? If im not mistaken, If I configured my trunk using fxo port and used the trunk in my outbound route, that’s should be it right?
Yes, just having NXXXXXX should be fine for 7 digit dialing. It will not match anything that is not 7 digits (Long Distance, International, 911…)
I assume the Dahdi Module of 2.8 refers to the FreePBX module, not the Dadhi Driver, correct? The FreePBX update should not impact anything as far as I know. There should be some release notes accessible via the module admin.
For outside an outside call, we have just been working on that. Setup the trunk and setup the outbound route. Unless you have custom contexts installed, it should be pretty straight forward.
I really can’t seem to figure out what’s wrong, so I decided to re-install everything. After I finished all the necessary installation, I’ve come up with this error.
retrieve_conf failed to sym link:
/etc/asterisk/chan_dahdi.conf from dahdiconfig/etc
This can result in FATAL failures to your PBX. If the target file exists and not identical, the symlink will not occur and you should rename the target file to allow the automatic sym link to occur and remove this error, unless this is an intentional customization.
Added 10 seconds ago
(retrieve_conf.SYMLINK)
All in-bounds call are working well, but still the outside call is not working. Does the error above caused the failure of outbound calls?
[quote=“dalenoll”]
All circuits busy indicates that there are no trunks available for that route, not that the far end is busy. If you defined your trunk(s) and have the route set to use those trunk(s), then perhaps the problem is at a lower level such as the configuration of the card (/etc/dahdi/system.confg, /etc/dahdi/modules.conf, /etc/asterisk/chan_dahdi.conf or /etc/asterisk/dahdi-channels.conf).[/quote]
I am not sure what FreePBX is trying to do with the dahdi config files. I know that FreePBX uses a ‘wrapper’ version of /etc/asterisk/chan_dahdi.conf which then ‘includes’ other files, such as chan_dahdi_additional.conf. But I am not sure what symbolic link it is trying to create.
This may also be something AsteriskNow specific, I do not use that distribution. Perhaps you should post the new error in the AsteriskNOW - FreePBX forum.
It is interesting that incoming works but outgoing does not. Can you post the output of the AsteriskCLI (core set verbose 3) when you attempt to make the outgoing call. Perhaps there is some information there that will make the issue clear.
[quote=“dalenoll”]
It is interesting that incoming works but outgoing does not. Can you post the output of the AsteriskCLI (core set verbose 3) when you attempt to make the outgoing call. Perhaps there is some information there that will make the issue clear.[/quote]
Wait, I am pertaining to inbound calls between extensions to extension, so that means fxs ports are working. Even with the softphones.