Invalid number cisco spa508g

Hi, I looked through the forums and Google but could not find an answer so I thought I’d ask here. I hope I’m in the right category? I setup Pbx in a flash Green and purchased service through the piaf Vitelity link everything was going good set everything up in freepbx then made calls through yate client softphone on my pc but cannot configure my Cisco spa508g phone from my previous provider that I reset. I manually set it up through the Cisco web interface sip, ipaddress of my local piaf box, userid, password etc… the phone lights up green has a dial tone shows up in freepbx as 2 ip phones online yate ext. and my cisco ext. but the phone won’t dial out it just says invalid number on the phone display after I dial the last digit and it gives me a busy tone. Thought maybe it was the phones dialplan tried to edit the default dialplan in the phone cisco web config tried (<:1765>xxxxxxxS0) I live in the USA but still invalid. My box is an old computer that is local that is connected to a switch on my modem/router

Thanks in advance for any help you can give to me.
Robert

Not to appear dense, but what is this (specifically, the 1765 thing?

I’m pretty sure that you don’t need anything by the 10 digit dial plan.

Hi thanks for your help it was to auto insert my area code 765 on local calls just trying different things loosing my mind.

Tech savvy but not a networker.

Thanks again,
Robert

OK.

If you have command line access to the server, try this:

# tail -F /var/log/asterisk/full >> /root/piaf.log

This will start up an extract of the current server log.

Dial a number. Wait until you get an indication the call failed.

Hit ^C.

Look at the log extract in /root/piaf.log

The number your phone sent will tell you a lot about what the phone is doing. If you can’t get if figured out from there, post the log here.

The more detail you provide the more likely you are to get someone looking that can help you.

Dave,

Thanks for your help I input the command that you supplied into the command line of the server itself but was unable to create the log file. I get error s^c: Substitution Failed when inputing caret and c after the call drops but I am a little closer I called my cisco hardphone from my softphone and my extension rang and I can receive incoming calls to it via my ringgroup but I still cannot make outgoing calls. it still says invalid number??? If that gives you anymore clues? I have reset the phone and haven’t configured anything in asterisk only in freepbx because they say it get over written just configured extensions and passwords and outbound routes etc… but no specific config files for the hardphone do I need those? Just manually in the web gui of the phone. Just trying to provide as much detail as I can. I think my freepbx is version 11 something piaf green as stated before.

Thanks for everything,
Robert

Honestly, the answer to your question is in the logs.

tail -F /var/log/asterisk/full >> /root/piaf.log
^C is “old style” for Control-C.

Log into your server as ‘root’. Type the tail command above. Once the tail program is running, nothing else will happen on the screen until you hit control-C. I’m going to assume you included the “#” in the command as well.

The point of the command was to get an extract of the /var/log/asterisk/full log so that you didn’t try to post thousands and thousands of lines of output from the asterisk program.

The information your phone is sending is there in the logs. If you can’t help us by showing us what’s in the logs, I doubt anyone is going to be able to psychically connect to your phone and help you.

try replacing

tail -F /var/log/asterisk/full >> /root/piaf.log

with

tailf /var/log/asterisk/full |tee -a /root/piaf.log

it will do the same thing without buffering but with visual feedback.

Thanks Dave and Dicko,

Here is a copy of the log file
Thanks Again,
Robert

[2015-02-13 13:09:28] VERBOSE[15970][C-0000000c] pbx.c: – Executing [[email protected]:1] Macro(“SIP/108-00000007”, “hangupcall,”) in new stack
[2015-02-13 13:09:28] VERBOSE[15970][C-0000000c] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/108-00000007”, “1?theend”) in new stack
[2015-02-13 13:09:28] VERBOSE[15970][C-0000000c] pbx.c: – Goto (macro-hangupcall,s,3)
[2015-02-13 13:09:28] VERBOSE[15970][C-0000000c] pbx.c: – Executing [[email protected]:3] ExecIf(“SIP/108-00000007”, “0?Set(CDR(recordingfile)=)”) in new stack
[2015-02-13 13:09:28] VERBOSE[15970][C-0000000c] pbx.c: – Executing [[email protected]:4] Hangup(“SIP/108-00000007”, “”) in new stack
[2015-02-13 13:09:28] VERBOSE[15970][C-0000000c] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/108-00000007’ in macro ‘hangupcall’
[2015-02-13 13:09:28] VERBOSE[15970][C-0000000c] pbx.c: == Spawn extension (macro-dial-one, h, 1) exited non-zero on ‘SIP/108-00000007’
[2015-02-13 13:09:28] VERBOSE[15970][C-0000000c] app_macro.c: == Spawn extension (macro-dial-one, s, 43) exited non-zero on ‘SIP/108-00000007’ in macro ‘dial-one’
[2015-02-13 13:09:28] VERBOSE[15970][C-0000000c] app_macro.c: == Spawn extension (macro-exten-vm, s, 16) exited non-zero on ‘SIP/108-00000007’ in macro ‘exten-vm’
[2015-02-13 13:09:28] VERBOSE[15970][C-0000000c] pbx.c: == Spawn extension (from-internal, 107, 2) exited non-zero on ‘SIP/108-00000007’
[2015-02-13 14:41:49] VERBOSE[1909][C-0000000d] netsock2.c: == Using SIP RTP TOS bits 184
[2015-02-13 14:41:49] VERBOSE[1909][C-0000000d] netsock2.c: == Using SIP RTP CoS mark 5
[2015-02-13 14:41:49] NOTICE[1909][C-0000000d] chan_sip.c: Call from ‘107’ (192.168.1.99:5060) to extension ‘17656493111’ rejected because extension not found in context ‘fxsgroup’.

what is in the fxsgroup context?

rasterisk -x ‘dialplan show fxsgroup’

don’t know? I was reading cisco docs online and they said to choose fsxgroup? I changed it for from-internal the same as my softphone extension and it works!!! Should I have to update the dialplan inside the web gui of the phone or if I leave it blank will it take the dialplan from freepbx

It is only an extension to FreePBX which by default are within from-internal’s includes, so no changes necessary on the phone.

rasterisk -x ‘dialplanshow [email protected]

to see the whole shebang.