Cant dial out , but can dial in

Hi,

After i started with the configuration of my elastix box i was able to make outgoing calls but no incoming calls.
Now i tested some parameters and now i can get incoming calls but when i try to make an outgoing call i get the message :

xecuting [[email protected]:2] Playback(“SIP/231-09a02cb8”, “pls-try-call-later|noanswer”) in new stack

.

My Trunkconfig is :

outgoing :

username=0341xxxxxx
type=friend
secret=XXXXX
host=213.148.136.2 (sip.qsc.de)
context=from-trunk
insecure=very
allow=ulaw&alaw&all
disallow=gsm

incoming:
type=friend
secret=XXXXXX
context=from-trunk
canreinvite=no
qualify=no
username=0341xxxxxx

dialpattern in the outgoing route is
XXX.

incoming is ANY DID / ANY CID

output from CLI when i try to make an outgoing call is :

-- Executing [[email protected]:1] Macro("SIP/231-09976bf8", "user-callerid|SKIPTTL|") in new stack
-- Executing [[email protected]:1] NoOp("SIP/231-09976bf8", "user-callerid: device 231") in new stack
-- Executing [[email protected]:2] Set("SIP/231-09976bf8", "AMPUSER=231") in new stack
-- Executing [[email protected]:3] GotoIf("SIP/231-09976bf8", "0?report") in new stack
-- Executing [[email protected]:4] ExecIf("SIP/231-09976bf8", "1|Set|REALCALLERIDNUM=231") in new stack
-- Executing [[email protected]:5] NoOp("SIP/231-09976bf8", "REALCALLERIDNUM is 231") in new stack
-- Executing [[email protected]:6] Set("SIP/231-09976bf8", "AMPUSER=231") in new stack
-- Executing [[email protected]:7] Set("SIP/231-09976bf8", "AMPUSERCIDNAME=chris") in new stack
-- Executing [[email protected]:8] GotoIf("SIP/231-09976bf8", "0?report") in new stack
-- Executing [[email protected]:9] Set("SIP/231-09976bf8", "AMPUSERCID=231") in new stack
-- Executing [[email protected]:10] Set("SIP/231-09976bf8", "CALLERID(all)="chris" <231>") in new stack
-- Executing [[email protected]:11] Set("SIP/231-09976bf8", "REALCALLERIDNUM=231") in new stack
-- Executing [[email protected]:12] ExecIf("SIP/231-09976bf8", "0|Set|CHANNEL(language)=") in new stack
-- Executing [[email protected]:13] NoOp("SIP/231-09976bf8", "TTL:  ARG1: SKIPTTL") in new stack
-- Executing [[email protected]:14] GotoIf("SIP/231-09976bf8", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [[email protected]:23] NoOp("SIP/231-09976bf8", "Using CallerID "chris" <231>") in new stack
-- Executing [[email protected]:2] Set("SIP/231-09976bf8", "_NODEST=") in new stack
-- Executing [[email protected]:3] Macro("SIP/231-09976bf8", "record-enable|231|OUT|") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/231-09976bf8", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [[email protected]:4] AGI("SIP/231-09976bf8", "recordingcheck|20080815-115539|1218794139.80") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck

recordingcheck|20080815-115539|1218794139.80: Outbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:5] NoOp(“SIP/231-09976bf8”, “No recording needed”) in new stack
– Executing [[email protected]:4] Macro(“SIP/231-09976bf8”, “dialout-trunk|2|01783530733||”) in new stack
– Executing [[email protected]:1] Set(“SIP/231-09976bf8”, “DIAL_TRUNK=2”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/231-09976bf8”, “0|Authenticate|”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/231-09976bf8”, “0?disabletrunk|1”) in new stack
– Executing [[email protected]:4] Set(“SIP/231-09976bf8”, “DIAL_NUMBER=01783530733”) in new stack
– Executing [[email protected]:5] Set(“SIP/231-09976bf8”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [[email protected]:6] Set(“SIP/231-09976bf8”, “GROUP()=OUT_2”) in new stack
– Executing [[email protected]:7] GotoIf(“SIP/231-09976bf8”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/231-09976bf8”, “0?skipoutcid”) in new stack
– Executing [[email protected]:10] Set(“SIP/231-09976bf8”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [[email protected]:11] Macro(“SIP/231-09976bf8”, “outbound-callerid|2”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/231-09976bf8”, “1?start”) in new stack
– Goto (macro-outbound-callerid,s,3)
– Executing [[email protected]:3] NoOp(“SIP/231-09976bf8”, “REALCALLERIDNUM is 231”) in new stack
– Executing [[email protected]:4] GotoIf(“SIP/231-09976bf8”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,9)
– Executing [[email protected]:9] Set(“SIP/231-09976bf8”, “USEROUTCID=”) in new stack
– Executing [[email protected]:10] Set(“SIP/231-09976bf8”, “EMERGENCYCID=”) in new stack
– Executing [[email protected]:11] Set(“SIP/231-09976bf8”, “TRUNKOUTCID=03411492737”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/231-09976bf8”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,16)
– Executing [[email protected]:16] GotoIf(“SIP/231-09976bf8”, “0?usercid”) in new stack
– Executing [[email protected]:17] Set(“SIP/231-09976bf8”, “CALLERID(all)=03411492737”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/231-09976bf8”, “1?report”) in new stack
– Goto (macro-outbound-callerid,s,22)
– Executing [[email protected]:22] NoOp(“SIP/231-09976bf8”, “CallerID set to “” <03411492737>”) in new stack
– Executing [[email protected]:12] AGI(“SIP/231-09976bf8”, “fixlocalprefix”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
– AGI Script fixlocalprefix completed, returning 0
– Executing [[email protected]:13] Set(“SIP/231-09976bf8”, “OUTNUM=01783530733”) in new stack
– Executing [[email protected]:14] Set(“SIP/231-09976bf8”, “custom=SIP/tom737”) in new stack
– Executing [[email protected]:15] GotoIf(“SIP/231-09976bf8”, “1?gocall”) in new stack
– Goto (macro-dialout-trunk,s,17)
– Executing [[email protected]:17] Macro(“SIP/231-09976bf8”, “dialout-trunk-predial-hook|”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/231-09976bf8”, “0?bypass|1”) in new stack
– Executing [[email protected]:19] GotoIf(“SIP/231-09976bf8”, “0?customtrunk”) in new stack
– Executing [[email protected]:20] Dial(“SIP/231-09976bf8”, “SIP/tom737/01783530733|300|”) in new stack
– Called tom737/01783530733
– SIP/tom737-09a22218 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [[email protected]:21] Goto(“SIP/231-09976bf8”, “s-CONGESTION|1”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,1)
– Executing [[email protected]:1] GotoIf(“SIP/231-09976bf8”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,3)
– Executing [[email protected]:3] NoOp(“SIP/231-09976bf8”, “TRUNK Dial failed due to CONGESTION - failing through to other trunks”) in new stack
– Executing [[email protected]:5] Macro(“SIP/231-09976bf8”, “outisbusy|”) in new stack
– Executing [[email protected]:1] Playback(“SIP/231-09976bf8”, “all-circuits-busy-now|noanswer”) in new stack

can anyone give mit a tip or hint where i have to look that i can fix this ?

My small experience of trunk configuration is that one small thing wrong can result in the dreaded “All circuits busy”. It also seems to be provider specific - so what works with one provider won’t necessarily work with another.

Does this thread help:

http://www.ip-phone-forum.de/showthread.php?t=71954

My German isn’t good enough to understand it, but there are some sample configs there.

I am experiencing similar symptoms as tkbeat.

I don’t know if our issues are related but I can’t get incomming or outgoing to work at all.

That is until I physical swap the FROM-PSTN and FROM-INTERNAL cables. Once the physical swap is changed (so all PSTN calls are actually coming from the internal context) - I get the ‘all circuits are busy’ on outbound calls, but ‘PSTN incoming’ works fine. (I’m still experimenting.)

(Trixbox 2.6.1.2, Sangoma A104d, Asterisk 1.4.20-1, zaptel-1.4.10.1)