Still having problems with inbound calls

I am still having problems with inbound calls. I can call and the call is rejected, then, call right back, from the same phone and the call will go through. The log snippit I am posting below is an example of that. I called, The very first line is where the call is rejected by Asterisk claiming that the extension doesn’t exist. I then called right back and starting with the second line, my call was properly processed. I greatly appreciate any help.

Thanks

[Jun 17 11:20:21] NOTICE[803]: chan_sip.c:14033 handle_request_invite: Call from ‘’ to extension ‘17772744835’ rejected because extension not found.
– Executing [17772744835@incoming_calls:1] Answer(“SIP/17772744835-0876f9c8”, “”) in new stack
– Executing [17772744835@incoming_calls:2] Wait(“SIP/17772744835-0876f9c8”,“1”) in new stack
– Executing [17772744835@incoming_calls:3] Playback(“SIP/17772744835-0876f9c8”, “at-tone-time-exactly”) in new stack
– <SIP/17772744835-0876f9c8> Playing ‘at-tone-time-exactly’ (language ‘en’)
– Executing [17772744835@incoming_calls:4] SayUnixTime(“SIP/17772744835-0876f9c8”, “||IMp”) in new stack
– <SIP/17772744835-0876f9c8> Playing ‘digits/11’ (language ‘en’)
– <SIP/17772744835-0876f9c8> Playing ‘digits/20’ (language ‘en’)
– <SIP/17772744835-0876f9c8> Playing ‘digits/a-m’ (language ‘en’)
– Executing [17772744835@incoming_calls:5] Playback(“SIP/17772744835-0876f9c8”, “beep”) in new stack
– <SIP/17772744835-0876f9c8> Playing ‘beep’ (language ‘en’)
– Executing [17772744835@incoming_calls:6] Playback(“SIP/17772744835-0876f9c8”, “temperature”) in new stack
– <SIP/17772744835-0876f9c8> Playing ‘temperature’ (language ‘en’)
– Executing [17772744835@incoming_calls:7] AGI(“SIP/17772744835-0876f9c8”,“gettemp.pl”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/gettemp.pl
– <SIP/17772744835-0876f9c8> Playing ‘digits/70’ (language ‘en’)
– <SIP/17772744835-0876f9c8> Playing ‘digits/8’ (language ‘en’)
– AGI Script gettemp.pl completed, returning 0
– Executing [17772744835@incoming_calls:8] Playback(“SIP/17772744835-0876f9c8”, “degrees”) in new stack
– <SIP/17772744835-0876f9c8> Playing ‘degrees’ (language ‘en’)
– Executing [17772744835@incoming_calls:9] Wait(“SIP/17772744835-0876f9c8”,“1”) in new stack
– Executing [17772744835@incoming_calls:10] Playback(“SIP/17772744835-0876f9c8”, “wind”) in new stack
– <SIP/17772744835-0876f9c8> Playing ‘wind’ (language ‘en’)
– Executing [17772744835@incoming_calls:11] Playback(“SIP/17772744835-0876f9c8”, “speed”) in new stack
– <SIP/17772744835-0876f9c8> Playing ‘speed’ (language ‘en’)
– Executing [17772744835@incoming_calls:12] Playback(“SIP/17772744835-0876f9c8”, “is”) in new stack
– <SIP/17772744835-0876f9c8> Playing ‘is’ (language ‘en’)
– Executing [17772744835@incoming_calls:13] AGI(“SIP/17772744835-0876f9c8”,“getwind.pl”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/getwind.pl
– <SIP/17772744835-0876f9c8> Playing ‘digits/3’ (language ‘en’)
– AGI Script getwind.pl completed, returning 0
– Executing [17772744835@incoming_calls:14] Playback(“SIP/17772744835-0876f9c8”, “miles-per-hour”) in new stack
– <SIP/17772744835-0876f9c8> Playing ‘miles-per-hour’ (language ‘en’)
– Executing [17772744835@incoming_calls:15] Playback(“SIP/17772744835-0876f9c8”, “goodbye”) in new stack
– <SIP/17772744835-0876f9c8> Playing ‘goodbye’ (language ‘en’)
– Executing [17772744835@incoming_calls:16] Wait(“SIP/17772744835-0876f9c8”, “1”) in new stack
– Executing [17772744835@incoming_calls:17] Hangup(“SIP/17772744835-0876f9c8”, “”) in new stack
== Spawn extension (incoming_calls, 17772744835, 17) exited non-zero on ‘SIP/17772744835-0876f9c8’

can u post ur extensions.conf file?

I know this is probably more than you want but I was afraid of cutting out something that might be important to the diagnosis… As you can see from my sip snippit below, inbound calls on the Callcentric line should use the incoming_calls context unless I’m missing something that is. :wink: The incoming_calls context is set up to just announce the time, temperature and windspeed then hangup.

SIP.CONF Snippit

[callcentric]
type=friend
context=incoming_calls
host=callcentric.com
username=17772744835
secret=supersecretpassword
fromuser=17772744835
fromdomain=callcentric.com
insecure=port,invite

EXTENSIONS.CONF

[globals]

; .........................................................
; Peers
; .........................................................

VOICEPULSE_GATEWAY_OUT_A=to-voicepulse
VOICEPULSE_GATEWAY_OUT_B=to-voicepulse-backup
VOICEPULSE_GATEWAY_OUT_C=to-voicepulse-legacy
MONITOR_EXEC=/usr/local/bin/wavIn2ogg.sh

[general]
autofallthrough=yes

[default]

exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

[incoming_calls]

exten => 17772744835,1,Answer
exten => 17772744835,n,Wait(1)
exten => 17772744835,n,Playback(at-tone-time-exactly)
exten => 17772744835,n,SayUnixTime(,IMp)
exten => 17772744835,n,Playback(beep)
exten => 17772744835,n,Playback(temperature)
exten => 17772744835,n,AGI(gettemp.pl)
exten => 17772744835,n,Playback(degrees)
exten => 17772744835,n,Wait(1)
exten => 17772744835,n,Playback(wind)
exten => 17772744835,n,Playback(speed)
exten => 17772744835,n,Playback(is)
exten => 17772744835,n,AGI(getwind.pl)
exten => 17772744835,n,Playback(miles-per-hour)
exten => 17772744835,n,Playback(goodbye)
exten => 17772744835,n,Wait(1)
exten => 17772744835,n,Hangup

; End CallCentric incoming

[outgoing_calls]
exten => _71777.,1,NoOp()
exten => _71777.,n,Dial(SIP/callcentric/${EXTEN:1})
exten => _91NXXNXXXXXX,1,Set(CALLERID(num)=19282682902)
exten => _91NXXNXXXXXX,2,Set(CALLERID(name)=19282682902)
exten => _91NXXNXXXXXX,n,NoOp(SIPCALLID: ${SIPCALLID})
exten => _91NXXNXXXXXX,n,Dial(SIP/+${EXTEN:1}@${VOICEPULSE_GATEWAY_OUT_A})
exten => _91NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN:1}|GatewayB)
exten => _91NXXNXXXXXX,n(GatewayB),Dial(SIP/+${EXTEN:1}@${VOICEPULSE_GATEWAY_OUT_B})
exten => _91NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN:1}|GatewayC)
exten => _91NXXNXXXXXX,n(GatewayC),Dial(SIP/${EXTEN:1}@${VOICEPULSE_GATEWAY_OUT_C})

[internal]
include => outgoing_calls
; Frank’s Extension
exten => 101,1,Verbose(1|Extension 101)
exten => 101,n,Dial(SIP/101,30)
exten => 101,n,VoiceMail(101@default)
exten => 101,n,PlayBack(vm-goodbye)
exten => 101,n,HangUp()

; Laura’s Extension
exten => 102,1,Verbose(1|Extension 102)
exten => 102,n,Dial(SIP/102,30)
exten => 102,n,VoiceMail(102@default)
exten => 102,n,PlayBack(vm-goodbye)
exten => 102,n,Hangup()

; this conference room is not recorded, for preparations
exten => 151,1,Answer
exten => 151,2,Wait(1)
exten => 151,3,Meetme(151,s)

; this conference room records automatically
exten => 152,1,Answer
exten => 152,2,Wait(1)
exten => 152,3,Set(CALLFILENAME=podcast_X_${CALLERID(name)}-${STRFTIME(${EPOCH},%Y%m%d-%H%M%S)})
exten => 152,4,Monitor(wav,${CALLFILENAME},m)
exten => 152,5,Meetme(152,s)

; Voicemail Retrieval
exten => 199,1,VoicemailMain(${CALLERID(num)},s)
exten => 199,2,Hangup

exten => *66,1,Answer
exten => *66,2,Playback(national-weather-service)
exten => *66,n,Wait(1)
exten => *66,n,System(/usr/bin/getweather)
exten => *66,n,Wait(1)
exten => *66,n,Playback(/tmp/weather)
exten => 66,n,System(rm /tmp/weather. -f)
exten => *66,n,Playback(goodbye)
exten => *66,n,Hangup

exten => *6,1,Answer
exten => *6,n,Wait(1)
exten => *6,n,Playback(at-tone-time-exactly)

exten => *6,n,SayUnixTime(,IMp)
exten => *6,n,Playback(beep)
exten => *6,n,Playback(temperature)
exten => *6,n,AGI(gettemp.pl)
exten => *6,n,Playback(degrees)
exten => *6,n,Wait(1)
exten => *6,n,Playback(wind)
exten => *6,n,Playback(speed)
exten => *6,n,Playback(is)
exten => *6,n,AGI(getwind.pl)
exten => *6,n,Playback(miles-per-hour)
exten => *6,n,Playback(goodbye)
exten => *6,n,Wait(1)
exten => *6,n,Hangup

exten => *3,1,Answer
exten => *3,n,Directory(default,internal,eb)
exten => *3,n,Hangup

; Test Echo App
exten => 500,1,Verbose(1|Echo test application)
exten => 500,n,SayUnixTime(,IMp)
exten => 500,n,Echo()
exten => 500,n,Hangup()

[phones]
include => internal

Did you ever get to the bottom of this issue?

I have the same problem on one of my boxes.

I have a Grandstream GXW4108 SIP-PSTN gateway that forwards all incoming calls to extension 444. I can call it 10 times and 3 out of 10 times it will fail with the error “NOTICE[3606]: chan_sip.c:13664 handle_request_invite: Call from ‘699’ to extension ‘444’ rejected because extension not found.” The other 7 times the system works perfectly.

If you have hit on what is casuing this, please pass it along!