To outgoing call if you don’t define one prefix you will dial as normal, in this case there is no prefix so you can dial normal.
Now if you can enter in the server and open a console with asterisk -r and paste all you see when dial
To outgoing call if you don’t define one prefix you will dial as normal, in this case there is no prefix so you can dial normal.
Now if you can enter in the server and open a console with asterisk -r and paste all you see when dial
[Jan 12 09:21:08] VERBOSE[5104] logger.c: – Executing [79874573@from-internal:1] ResetCDR(“SIP/20-0881dc20”, “”) in new stack
[Jan 12 09:21:08] VERBOSE[5104] logger.c: – Executing [79874573@from-internal:2] NoCDR(“SIP/20-0881dc20”, “”) in new stack
[Jan 12 09:21:08] VERBOSE[5104] logger.c: – Executing [79874573@from-internal:3] Wait(“SIP/20-0881dc20”, “1”) in new stack
[Jan 12 09:21:09] VERBOSE[5104] logger.c: – Executing [79874573@from-internal:4] Playback(“SIP/20-0881dc20”, “silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer”) in new stack
[Jan 12 09:21:09] VERBOSE[5104] logger.c: – <SIP/20-0881dc20> Playing ‘silence/1’ (language ‘en’)
[Jan 12 09:21:10] VERBOSE[5104] logger.c: – <SIP/20-0881dc20> Playing ‘cannot-complete-as-dialed’ (language ‘en’)
[Jan 12 09:21:13] VERBOSE[5104] logger.c: – <SIP/20-0881dc20> Playing ‘check-number-dial-again’ (language ‘en’)
[Jan 12 09:21:15] VERBOSE[5104] logger.c: – Executing [79874573@from-internal:5] Wait(“SIP/20-0881dc20”, “1”) in new stack
[Jan 12 09:21:16] VERBOSE[5104] logger.c: – Executing [79874573@from-internal:6] Congestion(“SIP/20-0881dc20”, “20”) in new stack
[Jan 12 09:21:16] VERBOSE[5104] logger.c: == Spawn extension (from-internal, 79874573, 6) exited non-zero on ‘SIP/20-0881dc20’
[Jan 12 09:21:16] VERBOSE[5104] logger.c: – Executing [h@from-internal:1] Macro(“SIP/20-0881dc20”, “hangupcall”) in new stack
[Jan 12 09:21:16] VERBOSE[5104] logger.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/20-0881dc20”, “1?skiprg”) in new stack
[Jan 12 09:21:16] VERBOSE[5104] logger.c: – Goto (macro-hangupcall,s,4)
[Jan 12 09:21:16] DEBUG[5104] app_macro.c: Executed application: GotoIf
[Jan 12 09:21:16] VERBOSE[5104] logger.c: – Executing [s@macro-hangupcall:4] GotoIf(“SIP/20-0881dc20”, “1?skipblkvm”) in new stack
[Jan 12 09:21:16] VERBOSE[5104] logger.c: – Goto (macro-hangupcall,s,7)
[Jan 12 09:21:16] DEBUG[5104] app_macro.c: Executed application: GotoIf
[Jan 12 09:21:16] VERBOSE[5104] logger.c: – Executing [s@macro-hangupcall:7] GotoIf(“SIP/20-0881dc20”, “1?theend”) in new stack
[Jan 12 09:21:16] VERBOSE[5104] logger.c: – Goto (macro-hangupcall,s,9)
[Jan 12 09:21:16] DEBUG[5104] app_macro.c: Executed application: GotoIf
[Jan 12 09:21:16] VERBOSE[5104] logger.c: – Executing [s@macro-hangupcall:9] Hangup(“SIP/20-0881dc20”, “”) in new stack
[Jan 12 09:21:16] VERBOSE[5104] logger.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/20-0881dc20’ in macro ‘hangupcall’
[Jan 12 09:21:16] VERBOSE[5104] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/20-0881dc20’
[Jan 12 09:21:28] VERBOSE[5105] logger.c: == Parsing ‘/etc/asterisk/manager.conf’: [Jan 12 09:21:28] VERBOSE[5105] logger.c: Found
[Jan 12 09:21:28] VERBOSE[5105] logger.c: == Parsing ‘/etc/asterisk/manager_additional.conf’: [Jan 12 09:21:28] VERBOSE[5105] logger.c: Found
[Jan 12 09:21:28] VERBOSE[5105] logger.c: == Parsing ‘/etc/asterisk/manager_custom.conf’: [Jan 12 09:21:28] VERBOSE[5105] logger.c: Found
[Jan 12 09:21:28] VERBOSE[5105] logger.c: == Manager ‘admin’ logged on from 127.0.0.1
When I made a change in the GUI I cant see the button to apply, so enter in the CLI and perform a reload command and dahdi restart too all in the asterisk CLI.
If you still can’t dial out run this command in your asterisk CLI:
And paste the output please.
Hi
Sorry for the late reply
I just received my 8 port fxo card, but I am getting confused on how to check if the card is working and how to set the outbound and inbound calls. I will use just sip phones.
I am using asterisk 1.4.24 with free pbx 2.7 through the web interface.
Thanks for your help
Justin
Configure inbound routes in FreePBX with the last 4 numbers of the line number, for each line in your fxo card, and then set destinations to your IVR or extensions; or You can set any DID for send the call to the same IVR or extensions, just leave blank the field for DID.
You need to configure your zaptel/dahdi config files first.
Thanks for your help
and also when i try #dahdi_hardware -vvv nothing happens, is this ok or a bad sign ?
Bad sign if you install dahdi and a Digital or Analog Card, but if you install zaptel try run zaptel_hardware -vvvv.
pls can you be so kind to access my box and check it out for me as am desperate ?
I will pm yo my new ip and drivers rcvd with the card
Is it ok ?
Thanks
also should i connect the power connector on the card if i am using it exclusively as an FXO ?
and I have another problem the internal extension work fine on the local network but from over the net (router and port configured) the phones ring but no voice is transferred
please can you help me out
thanks
justin
Nope, only for FXS
Need to specify Externhost or ExternIP or Stunserver, and the refresh time. And grant acces for rtp ports and sip port.
may pm you all the details and have a look pls ?
ok
ok sent let me know if you got it pls