SOLVED Help with POTS migration

[SipnEat]

Hey All -

Ok Im in a pickle. I have a new PIAF system setup using Asterisk 11, latest FreePBX. I have installed a Sangoma B600-DE and under the DAHDi settings a see ANALOG and the 4 FXO ports and the 1 FXS ports. I did notice that the card has a green light on the I FXS port and red on the FXO Ports on the physical card. Im not sure if that matters. Sorry never used one before.

The customer has the current three POTS lines going to a TalkSwitch, all working as expected. Im trying to move them over to the new PBX.

When I move two of the three lines over, and we call the main number we get a "Line is not in service" I believe. Moving them back to the talk switch, poof phones are ringing..

So, yes Im a PBX n00b, but I have been working is systems for well, lets just say a LONG time.. don't want to age myself, but my first computer was a Apple II+ .

So what am I asking... are there any specific things I should be looking for in the config? I have an inbound route, bound to the main DID, but the fact that its saying the line is down, makes me think there is something else going on here..

THANKS!!!

Brian

 

[SipnEat]

I did set the MAIN number to the analog channel, so I think that's good there..

I do have a SIPSTATION SIP trunk also setup (that's working), should I remove it? I do not see any errors in the log, but I do not see any info on DAHDi stuff on reboot, like "Hello here I am". Im assuming that the fact it shoes up with info when I look at the DHADi (connectivity --> DAHDi) that its there...

THANKS!

 

[SipnEat]

here is what I get

dahdi show channels:
Chan Extension Context Language MOH Interpret Blocked State Description
pseudo default default In Service
1 from-analog en default In Service
2 from-analog en default In Service
3 from-analog en default In Service
4 from-analog en default In Service
5 from-analog en default In Service

dahdi show channel 1 (the others are the same)
Channel: 1
Description:
File Descriptor: 11
Span: 1
Extension:
Dialing: no
Context: from-analog
Caller ID:
Calling TON: 0
Caller ID subaddress:
Caller ID name:
Mailbox: none
Destroy: 0
InAlarm: 1
Signalling Type: FXS Kewlstart
Radio: 0
Owner: <None>
Real: <None>
Callwait: <None>
Threeway: <None>
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Busy Detection: yes
Busy Count: 10
Busy Pattern: 0,0,0,0
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Gains (RX/TX): 0.00/0.00
Dynamic Range Compression (RX/TX): 0.00/0.00
DND: no
Echo Cancellation:
128 taps
(unless TDM bridged) currently OFF
Wait for dialtone: 0ms
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Onhook

 

[SipnEat]

Ok channel 5 is the FXS port, and not being used. So have not done much with it..

Few more details...

Im use only the FIRST phone number in the group, it is the main number though. I am not connecting the other could that be an issue?

Ok now for the n00b question.. Which log shows the call detail? The standard one that comes up when I go to LOGS seems to be it, but there is a pretty big list of logs.. if I can get a more specific and therefore a more details log ill send it out..

THANKS!

-b

 

[SipnEat]

Ok so do I need to do something special for an incoming route? When I EDIT the DAHDI trunk it states that its "In use by 1 route", the notes say its DefaultOut (my name) Sequence 2 (Its secondary from the SIP trunk). The SIP trunks say the same thing... so not sure if I need to add something to the DAHDi incoming..

thanks!

-b

 

[SipnEat]

UPDATE - Its all working now! I wish I could post "this is what the issue was, and a fix" but the fact is, it just started working. Im SURE I did something, or maybe the hardware check did it, not sure... but, moving the lines over works now... IVR, etc..

THANKS!