I just had this working, now I don’t know what I’ve done wrong…I get a phone call when I’m testing, but I am not hearing a sound clip when I receive the call. I just answer, and nothing is on the other end. I’ll paste /var/log/asterisk/message logs
[Mar 19 16:55:00] VERBOSE[14395] pbx_spool.c: – Attempting call on sip/voipvoip/myphonenumber for t@mycontext:1 (Retry 1)
[Mar 19 16:55:00] VERBOSE[14395] netsock.c: == Using SIP RTP CoS mark 5
[Mar 19 16:55:12] VERBOSE[14395] pbx.c: == Starting SIP/voipvoip-b6f69878 at mycontext,t,1 failed so falling back to exten ‘s’
[Mar 19 16:55:12] VERBOSE[14395] pbx.c: == Starting SIP/voipvoip-b6f69878 at mycontext,s,1 still failed so falling back to context ‘default’
[Mar 19 16:55:12] VERBOSE[14395] pbx.c: – Executing [s@default:1] Answer(“SIP/voipvoip-b6f69878”, “”) in new stack
[Mar 19 16:55:12] VERBOSE[14395] pbx.c: – Executing [s@default:2] Set(“SIP/voipvoip-b6f69878”, “TIMEOUT(digit)=5”) in new stack
[Mar 19 16:55:12] VERBOSE[14395] func_timeout.c: – Digit timeout set to 5.000
[Mar 19 16:55:12] VERBOSE[14395] pbx.c: – Executing [s@default:3] Set(“SIP/voipvoip-b6f69878”, “TIMEOUT(response)=10”) in new stack
[Mar 19 16:55:12] VERBOSE[14395] func_timeout.c: – Response timeout set to 10.000
[Mar 19 16:55:12] VERBOSE[14395] pbx.c: – Executing [s@default:4] Wait(“SIP/voipvoip-b6f69878”, “1”) in new stack
[Mar 19 16:55:13] VERBOSE[14395] pbx.c: – Executing [s@default:5] Playback(“SIP/voipvoip-b6f69878”, “/usr/share/asterisk/sounds/en/Test”) in new stack
[Mar 19 16:55:13] WARNING[14395] file.c: File /usr/share/asterisk/sounds/en/Test does not exist in any format
[Mar 19 16:55:13] WARNING[14395] file.c: Unable to open /usr/share/asterisk/sounds/en/Test (format 0x4 (ulaw)): No such file or directory
[Mar 19 16:55:13] WARNING[14395] app_playback.c: ast_streamfile failed on SIP/voipvoip-b6f69878 for /usr/share/asterisk/sounds/en/Test
[Mar 19 16:55:13] VERBOSE[14395] pbx.c: – Auto fallthrough, channel ‘SIP/voipvoip-b6f69878’ status is ‘UNKNOWN’
[Mar 19 16:55:13] NOTICE[14395] pbx_spool.c: Call completed to sip/voipvoip/myphonenumber
I do have a context in /etc/asterisk/extensions.conf called [mycontext]
[mycontext]
exten => t,1,Answer
exten => t,2,Set(TIMEOUT(digit)=5)
exten => t,3,Set(TIMEOUT(response)=10)
exten => t,4,Wait(1)
exten => t,5,Playback(name)
exten => t,6,Wait(1)
exten => t,7,Hangup
…and, name.gsm does exist in /usr/share/asterisk/sounds/en
The logs tell me that for some reason the context isn’t being executed. I’ve tested that the sound file works by listening to it. So, I know it isn’t that. I did have the letter ‘t’ in my context (the extension) set to ‘s’ before…but since that wasn’t working I change it to ‘t’ for testing purposes.
I’m lost.