Analog trunk question

Hello,

I have an analog card with two FXO ports where PSTN lines are connected.

Our main number is xxx-xxx-0050 and the next number in the hunt (set by our telco) is xxx-xxx-0051

I have one trunk, one inbound route and one outbound route setup.

I’m a little confused if I need to create two trunks (one for each line connected to each port on the FXO card) or if one trunk will handle both lines.

With only two lines, what would be the recommended way to handle outbound phone calls. Would specifying the second port to handle these calls going out, while the first port would be for the incoming (main phone number) calls?

I think I have something configured wrong because there are random times where I try to make an outbound call and it rings with no answer. Sometimes the call is completed and other times it just rings. I have tried this with my cell phone in hand while dialing out from the voip system.

The only thing that I have configured in the Outbound route is the dial pattern of “1NXXNXXXXXX”.

Sometimes it seems like I restart the dahdi & asterisk in the dahdi configuration and will complete the outbound calls for a short period of time.

Any help is appreciated!

basically a trunk is a group of one or more “lines”, you can refer to the lines as dahdi/1,dahdi/2 etc. but dahdi allows you to group them in the /etc/asterisk/chan_dahdi.conf and any “included” files file so you can refer to by “groups”, primarily dahdi/g0 = ascending, dahdi/G0 as descending, dahdi/r0 as circular hunt up and dahdi/R0 as circular hunt down , generally to prevent “glare” use a hunt group opposite of your incoming hunt for outgoing calls so for two lines well wired then use G0

It’s behavior is more fully described at:-

http://docs.tzafrir.org.il/dahdi-tools/

A quick guess knowing your previous history here, is that you are clean out of usable channels because you are incorrectly doing “hangup supervision” because cheap Adtrans don’t support it, you will need to set busydetect=yes etc. and wait a couple of minutes.

After trying various configuration changes, reading multiple resources on config files and experimenting I decided to change paths. I purchased a Grandstream UCM6104 to replace the pbx box I have built with the TM400 telephony card.

I had it setup within a 1/2 hour with all extensions setup. I copied the SIP passwords from the original pbx to the Grandstream so the users wouldn’t need to update the password in their soft/hard phones.

I must say that I’m pretty impressed with the small footprint and the similar but nicely laid out interface. Most importantly the issues that I was having with the analog lines not properly disconnecting are now working perfect. The price of the device was 260.00 or so on amazon.

Thanks for everyone’s input.