New user help on outgoing calls from TDM410

Hi, I have been trying to set up a simple system for over a week. I am a complete newby at Linux and Voip so it have been a steep learning curve. So far i have set up Asterisk 11.1.2 with the latest beta of FreePBX from scratch on CentOS 6.3 (I did this because the distro just would not install on my hardware for some reason). I have also managed to install a SIP connection and a TDM410 card giving me access to outside lines through VOIP or analogue using a Digium D40 and a Grandstream GPX2100.

Where i am completely stuck is on getting the incoming calls to work. Below is the error message I get:


Connected to Asterisk 11.1.2 currently running on voip (pid = 2941)
– Starting simple switch on ‘DAHDI/1-1’
[2013-01-15 21:39:24] WARNING[3823][C-00000015]: pbx.c:6167 __ast_pbx_run: Channel ‘DAHDI/1-1’ sent to invalid extension but no invalid handler: context,exten,priority=from_pstn,s,1
– Hanging up on ‘DAHDI/1-1’
– Hungup ‘DAHDI/1-1’

The system can dial out and sees the call coming in but cant route it for some reason that i am at a loss to. I have one incoming route that is completely open (no did/cid info) that i thought would just catch anything and route it to my phone extension 100. It does this for the Voip line no issue.

I set up the trunk as a Dahdi trunk with no CID info, no dial prefixes and using group 0 acsending mode.

The outbound route is setup with a dial prefix of 70 so for outgoing i can dial 70 for the analog line and 80 for the SIP line - both of these work fine.

I have been trying to mess with the context settings trying from internal, zaptel, pstn, dahdi in the dahdi.conf file but none of those made any difference

Some of the info on the setup

dahdi show channels

Chan Extension Context Language MOH Interpret Blocked State Description
pseudo default default In Service
1 from_pstn en default In Service
2 en default In Service
3 en default In Service
4 from-analog en default In Service


voip*CLI> dahdi show channel 1 (this is the connected FXO line)
Channel: 1
Description:
File Descriptor: 21
Span: 1
Extension:
Dialing: no
Context: from_pstn
Caller ID:
Calling TON: 0
Caller ID subaddress:
Caller ID name:
Mailbox: none
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner:
Real:
Callwait:
Threeway:
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): Offhook

Any help or a quick walkthrough of how to get this working would be greatly appreciated. I have gotten this far just by reading all the help people have given others on this site so am hoping you can put up with one more new guy.
If there is some other info i should pull down to help then let me know

You need to assign a Zap Channel DID for this.
Keep in mind that analog inbound lines do not include what number is being called.
You do this by attaching the channel to a Zap DID. So Channel = 1, DID = your incoming number. After this you can use your inbound routes to match on this number being called.
–MTLVoice

MTL is incorrect, this has nothing to do with Zap DID’s.

The context’s you probided ‘from_pstn’ is invalid, the correct contect is ‘from-pstn’

Please also correct channel 4

Thanks both for the replies. I will change the underscore as soon as i get home tonight.

For my understanding on DID’s:

If my incoming analog phone number to channel 1 is 016xxxxxxx
Then in Dahdi DID Channels I set channel 1 to 016xxxxxxx
Then on the inbound route i set the CID to 016xxxxxxx?

Hi, just to let you know - i changed the underscore to a dash and I got success. Amazing, thanks so much for spotting it for me.

Correct, that way if you have multiple analog lines connected to your system, you can distinguish between calls arriving on each line, and should you wish, do something depending on which line is ringing.

–MTLVoice

Thanks again for clarifying that one for me