Arrgghhh… No.
Let me go make a crossover cable and I will get back to you.
/smackinthehead
Dicko, you are the man!
I’m green on both spans and can make outgoing calls from the legacy PBX. Now I just have to get incoming working. I am still getting the number out of service message when I try to call in.
Well you are over the hard bit, now you need to explore your call routing, watch a call for the DID requested, it might be 10 or just the last 4 or 3 digits when you get it, think about using from-did-direct or even from-internal (yes quite safe for a PRI) for the g0 context, arrange for unresolved DID’s to go straight out an outbound route to g1 (probably matching 55555555XX where 55555555 (or appropriately less 
) are the Most Significant Digits of your PSTN numbers) other endpoints will be handled by your FreePBX/Asterisk thingy as you add them.
I’m not concerned about the DIDs right now, as I want it to work for a bit as a straight pass-through.
Currently I two DAHDI trunks set up, Windstream on Span 1 and Cerato on Span 2.
Span 1:
Trunk Name: Windstream
Outbound CallerID: (blank)
CID Options: Allow Any CID
Maximum Channels: (blank)
Asterisk Trunk Dial Options: (default) Override: (not checked)
Continue if Busy: (not checked)
Disable Trunk: (not checked)
Dialed Number Manipulation Rules: (no rules)
Outgoing Settings
DAHDI Trunks: Group 0 Ascending
Span 2:
Trunk Name: Cerato
Outbound CallerID: (blank)
CID Options: Allow Any CID
Maximum Channels: (blank)
Asterisk Trunk Dial Options: (default) Override: (not checked)
Continue if Busy: (not checked)
Disable Trunk: (not checked)
Dialed Number Manipulation Rules: (no rules)
Outgoing Settings
DAHDI Trunks: Group 0 Round Robin Ascending
I have one inbound route set for Any DID / Any CID set to Destination: Trunks / Cerato (dahdi)
I have one outbound route set for Any DID / Any CID set to Trunk Sequence 0 Windstream.
The system APPEARS to be running correctly, but I am sure I am FAR from done. Do you mind looking it over and seeing where I might be making some mistakes?
Thanks,
Scott
If it works then be happy, from an earlier post of yours:-
“adoption/migration to the FreePBX infrastructure will go very quickly and most, if not all, users will integrate and rely on the expanded feature sets / functionality.”
well, that is what you won’t be able to do until you wrap your head around DID’s and selective routing.
I spoke too soon…
Looks like I can get either incoming or outgoing working, but not both at the same time. Pulling my hair out trying to wrap my mind around the trunk group settings and the inbound/outbound rules…
The problem with 99% of the tutorials out there is that they seem to predominantly be about SIP and very few talk about incoming PRIs and pass-through to legacy PBXs…
As I said, you will need to selectively route both inbound and outbound calls from either trunk as appropriate. “Any CID/DID” routes will just not work for you, if you do that impeccably then adding SIP trunking for inbound/outbound calling will be seamless for internal or “legacy” extensions.
Ahh. I think I have an idea where to go with this now. I’ll keep trying, and thanks a lot for your advice!
Dicko:
I decided to take a step back and make sure I am doing things in the right order. If you don’t mind, can you see if I am on the correct track?
Step 1 was to configure each span of the T205P in the FreePBX box.
Span 1 is set up as follows:
Span 2 is setup as follows:
One of my issues is wrapping my head around channels and channel groups. According to the documentation I have on the PRI/T1 from Windstream, it is configured with voice on channels 1 through 15 and D channel on 24. We have a total of 36 DIDs assigned to us.
The legacy PBX is set up so that all extensions but two calling out use the main DID number for the organization. The fax extension is set up so that inbound and outbound calls on that DID only go to that extension and send the proper CLI information when calling out. The extension the fundraiser for our charitable foundation uses is set so that outbound CLI information is for that organization/department. Incoming calls on that DID go to the operator who has a different ring pattern assigned so she knows someone is calling into the charitable foundation, not the main organization.
So, here are my questions:
Are the trunks set up correctly for a straight bi-directional pass-through of the calls to and from the PRI to the legacy PBX?
What, if any, special configuration do I need to do to the groups in each trunk?
Thanks,
Scott
You have a fractional T1, your B channels need to be limited to 1-15 was that not obvious to you?, almost certainly using from-digital will cause you a lot more effort than needed, as I suggested use from-did-direct or from-internal, but If you can’t be bothered to watch an inbound call for what DID is being called as I have repeatedly suggested you do, then nor can I be bothered helping you any more 
Further, your clock source for the PSTN trunk needs to be the far end (1).
Dicko:
Thanks again, SO MUCH, for all your help. Believe me, anything I have done/tried is only from my position as a novice and not from ignoring your advice.
I am having difficulty finding a resource for the options available for use in the Groups Context settings. If I understand you, I should set Span 1’s Group 0 to Context: from-did-direct and Used Channels to 15. I did that, and it auto-created a group 1 for the remaining 8 channels. Do I ‘blank’ out the group settings for that and just ignore it?
As far as Span 2, which is connected to the legacy PBX via the turnaround cable (thanks again!), what is your suggestion as to the best way to configure the group(s)/Context?
Thanks,
Scott
Personally I would do it in the traditional dahdi setup and not rely on the FreePBX dahdi “helper” module, it will be less confusing. Basically work with two groups, 0 for the PSTN span (1-15) and 1 for the “legacy” span (24-47 or 24-39 depending on how your legacy PBX is set up, only you would be able to find out), without doubt the PBX trunk should be from-internal using from-digital only makes sense if your received DID’s are ten digits long and you don’t want to use the maximum advantage of your yet to be programmed endpoints, answer that question and stop guessing and you will be able to go further, I am not a mind reader.
Dicko:
Based upon my examination of the legacy PBX’s web interface configurations, the received DIDs from Windstream are 4 digits, so I am setting the PBX trunk (Span 2) to from-internal, like you suggested. It looks like the legacy PBX is also set up for 15 channels, so I will use the settings you suggest for that, as well.
Also, as I believe you suggested, I am setting up incoming routes for each of the DIDs. My only question is the destination setting. I had it set for Trunks : Cerato (the legacy PBX). Am I off base here?
Updated:
Is my thinking here messed up? Should I be setting the legacy PBX up as an extension and stop thinking of the trunk as the destination?
Thanks again,
Scott
I think I actually suggested what I would do before, you shouldn’t need any inbound routes defined.
If you have the PSTN trunk in the from-internal context and an outbound route to G1 (or g1 depending on how you want to do it) for a regex that matches your 36 DID’s (strange one that! ) pragmatically you could use 55XX where you replace 55 with what your real third and fourth least significant digits are, you will not be able to call the other 64 stations in your “common-block” though.
Only endpoints like extensions or IVR’s etc. that you have defined would then be handled by your FreePBX/Asterisk box as you add them within your 36 number range, everything else should fall through to your legacy box.
Feel free to expand your legacy PBX to use all 23 channels when it is working.
Dicko, you are a life-saver, a gentleman and a scholar…
Dicko:
Well, I have gone back and set everything as you suggested and all incoming calls are working and routing properly to the legacy PBX. DID to the main number is routing properly to the receptionist’s desk and DID to the fax machine is working as well.
Outgoing calls are another matter. All I get when I dial out of the legacy PBX is “The number you have dialed is not in service”. The logfile shows that it is an asterisk/FreePBX system message. It seems that no matter how I configure my outgoing rule I get this message.
Any thoughts?
What do your outbound routes look like?, what does the legacy box dial on the pbx to FreePBX trunk? Sooner or later you are going to have to venture into the Asterisk CLI so you can help yourself, you should post a log of a failed call.
Dicko:
Sorry for they delay. We are in election season and I have been tied up with that since the last post. That being said…
Being a DOS guy from the 80’s, you’d think I wouldn’t be this hesitant to dig into the CLI
I think I might be at the point were I need to contract with someone to talk me through this setup remotely. Is that something you would be interested in doing, or can you suggest anyone?
Thanks,
Scott
Schmooze , who host this whole kaboodle have a paid help system, I would prefer you to use that as self promotion is not appropriate here and I am much more expensive . If they can’t help then PM me.