TWO HT503 Got SIP response 486 "Busy Here" back from"

system · August 18, 2011, 11:54pm

look up my setting.

a first device HT503 (192.168.0.11) fxo registered ok in my asterisk server
a second device HT503 (192.168.0.12) fxo registered ok in my asterisk server

my first prefered outgoing route is 192.168.0.11 (pstn trunk)
my second route if first one is busy is 192.168.0.12 (pstn trunk)

look what I do

from any extension, I dial a pstn destination that connects ok (using route 192.168.0.11 with no problem)
while call 1 (using 192.168.0.11) is connected, from another extension I try to dial a diferent pstn destination that should connect using my second prefered trunk (192.168.0.12). what a sorprice, no success.

check errors in backgroung

command: tail -f /var/log/asterisk/full | grep response

[Aug 18 18:12:41] VERBOSE[3407] chan_sip.c: – Got SIP response 486 “Busy Here” back from 192.168.0.12

command: tail -f /var/log/asterisk/full

p(“SIP/8005001-00000092”, [Aug 18 18:18:06] VERBOSE[5066] pbx.c: [Aug 18 18:18:08] NOTICE[3411] [Aug 18 18:18:11] VERBOSE[3407] netsock.c: [Aug 18 18:18:11] VERBOSE[3407] netsock.c: [Aug 18 18:18:11] VERBOSE[5069] pbx.c: [Aug 18 18:18:11] VERBOSE[5069] pbx.c: [Aug 18 18:18:11] VERBOSE[5069] pbx.c: [Aug 18 18:18:11] VERBOSE[5069] pbx.c: [Aug 18 18:18:11] VERBOSE[5069] pbx.c: [Aug 18 18:18:11] VERBOSE[5069] pbx.c: [Aug 18 18:18:11] VERBOSE[5069] pbx.c: [Aug 18 18:18:11] VERBOSE[5069] pbx.c: [Aug 18 18:18:11] VERBOSE[5069] pbx.c: [Aug 18 18:18:11] VERBOSE[5069] pbx.c: [Aug 18 18:18:11] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] res_agi.c: [Aug 18 18:18:12] VERBOSE[5070] app_mixmonitor.c: [Aug 18 18:18:12] VERBOSE[5069] res_agi.c: [Aug 18 18:18:12] VERBOSE[5069] res_agi.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] pbx.c: [Aug 18 18:18:12] VERBOSE[5069] netsock.c: [Aug 18 18:18:12] VERBOSE[5069] netsock.c: “Setting Outbound Route MoH To: acc_1”) in new stack
– Executing [s@macro-setmusic:2] SetMusicOnHold(“SIP/8005001-00000092”, “acc_1”) in new stack
chan_iax2.c: Restricting registration for peer ‘109’ to 60 seconds (requested 300)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [2384375@from-internal:1] Macro(“SIP/110-00000093”, “user-callerid,SKIPTTL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/110-00000093”, “AMPUSER=110”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/110-00000093”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/110-00000093”, “1?Set(REALCALLERIDNUM=110)”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/110-00000093”, “AMPUSER=110”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/110-00000093”, “AMPUSERCIDNAME=110”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/110-00000093”, “0?report”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/110-00000093”, “AMPUSERCID=110”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/110-00000093”, “CALLERID(all)=“110” <110>”) in new stack
– Executing [s@macro-user-callerid:9] ExecIf(“SIP/110-00000093”, “1?Set(CHANNEL(language)=es)”) in new stack
– Executing [s@macro-user-callerid:10] GotoIf(“SIP/110-00000093”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [s@macro-user-callerid:19] NoOp(“SIP/110-00000093”, “Using CallerID “110” <110>”) in new stack
– Executing [2384375@from-internal:2] Set(“SIP/110-00000093”, “MOHCLASS=acc_1”) in new stack
– Executing [2384375@from-internal:3] Set(“SIP/110-00000093”, “_NODEST=”) in new stack
– Executing [2384375@from-internal:4] Macro(“SIP/110-00000093”, “record-enable,110,OUT,”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/110-00000093”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] ExecIf(“SIP/110-00000093”, “0?MacroExit()”) in new stack
– Executing [s@macro-record-enable:5] GotoIf(“SIP/110-00000093”, “0?Group:OUT”) in new stack
– Goto (macro-record-enable,s,15)
– Executing [s@macro-record-enable:15] GotoIf(“SIP/110-00000093”, “0?IN”) in new stack
– Executing [s@macro-record-enable:16] ExecIf(“SIP/110-00000093”, “0?MacroExit()”) in new stack
– Executing [s@macro-record-enable:17] NoOp(“SIP/110-00000093”, “Recording enable for 110”) in new stack
– Executing [s@macro-record-enable:18] Set(“SIP/110-00000093”, “CALLFILENAME=OUT110-20110818-181812-1313709491.147”) in new stack
– Executing [s@macro-record-enable:19] Goto(“SIP/110-00000093”, “record”) in new stack
– Goto (macro-record-enable,s,23)
– Executing [s@macro-record-enable:23] MixMonitor(“SIP/110-00000093”, “OUT110-20110818-181812-1313709491.147.wav,”) in new stack
– Executing [s@macro-record-enable:24] Set(“SIP/110-00000093”, “CDR(userfield)=audio:OUT110-20110818-181812-1313709491.147.wav”) in new stack
– Executing [s@macro-record-enable:25] MacroExit(“SIP/110-00000093”, “”) in new stack
– Executing [2384375@from-internal:5] Macro(“SIP/110-00000093”, “dialout-trunk,2,2384375,”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/110-00000093”, “DIAL_TRUNK=2”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/110-00000093”, “0?sub-pincheck,s,1”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/110-00000093”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/110-00000093”, “DIAL_NUMBER=2384375”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/110-00000093”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/110-00000093”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/110-00000093”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/110-00000093”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/110-00000093”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/110-00000093”, “outbound-callerid,2”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/110-00000093”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/110-00000093”, “0?Set(REALCALLERIDNUM=110)”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/110-00000093”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/110-00000093”, “USEROUTCID=110”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/110-00000093”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/110-00000093”, “TRUNKOUTCID=8005001”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/110-00000093”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] ExecIf(“SIP/110-00000093”, “1?Set(CALLERID(all)=8005001)”) in new stack
– Executing [s@macro-outbound-callerid:13] ExecIf(“SIP/110-00000093”, “1?Set(CALLERID(all)=110)”) in new stack
– Executing [s@macro-outbound-callerid:14] ExecIf(“SIP/110-00000093”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:15] ExecIf(“SIP/110-00000093”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [s@macro-dialout-trunk:12] ExecIf(“SIP/110-00000093”, “1?AGI(fixlocalprefix)”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
== Begin MixMonitor Recording SIP/110-00000093
== fixlocalprefix: Dialpattern NXXXXXX matched. 2384375 -> 2384375
– <SIP/110-00000093>AGI Script fixlocalprefix completed, returning 0
– Executing [s@macro-dialout-trunk:13] Set(“SIP/110-00000093”, “OUTNUM=2384375”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/110-00000093”, “custom=SIP/8005001”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/110-00000093”, “1?Set(DIAL_TRUNK_OPTIONS=M(setmusic^acc_1))”) in new stack
– Executing [s@macro-dialout-trunk:16] Macro(“SIP/110-00000093”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/110-00000093”, “”) in new stack
– Executing [s@macro-dialout-trunk:17] GotoIf(“SIP/110-00000093”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/110-00000093”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:19] Dial(“SIP/110-00000093”, “SIP/8005001/2384375,300,M(setmusic^acc_1)”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

[color=#FF0000][Aug 18 18:18:12] VERBOSE[5069] app_dial.c: – Called 8005001/2384375
[Aug 18 18:18:12] VERBOSE[3407] chan_sip.c: – Got SIP response 486 “Busy Here” back from 192.168.0.12
[Aug 18 18:18:12] VERBOSE[5069] app_dial.c: – SIP/8005001-00000094 is busy
[Aug 18 18:18:12] VERBOSE[5069] app_dial.c: == Everyone is busy/congested at this time (1:1/0/0)
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s@macro-dialout-trunk:20] NoOp(“SIP/110-00000093”, “Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s@macro-dialout-trunk:21] Goto(“SIP/110-00000093”, “s-BUSY,1”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Goto (macro-dialout-trunk,s-BUSY,1)
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s-BUSY@macro-dialout-trunk:1] NoOp(“SIP/110-00000093”, “Dial failed due to trunk reporting BUSY - giving up”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s-BUSY@macro-dialout-trunk:2] PlayTones(“SIP/110-00000093”, “busy”) in new stack
[Aug 18 18:18:12] WARNING[5069] translate.c: no samples for ulawtolin[/color]

[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s-BUSY@macro-dialout-trunk:3] Busy(“SIP/110-00000093”, “20”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] app_macro.c: == Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on ‘SIP/110-00000093’ in macro ‘dialout-trunk’
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: == Spawn extension (from-internal, 2384375, 5) exited non-zero on ‘SIP/110-00000093’
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [h@from-internal:1] Macro(“SIP/110-00000093”, “hangupcall”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/110-00000093”, “1?noautomon”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Goto (macro-hangupcall,s,3)
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s@macro-hangupcall:3] NoOp(“SIP/110-00000093”, “TOUCH_MONITOR_OUTPUT=”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s@macro-hangupcall:4] GotoIf(“SIP/110-00000093”, “1?skiprg”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Goto (macro-hangupcall,s,7)
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s@macro-hangupcall:7] GotoIf(“SIP/110-00000093”, “1?skipblkvm”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Goto (macro-hangupcall,s,10)
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s@macro-hangupcall:10] GotoIf(“SIP/110-00000093”, “1?theend”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Goto (macro-hangupcall,s,12)
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s@macro-hangupcall:12] Hangup(“SIP/110-00000093”, “”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] app_macro.c: == Spawn extension (macro-hangupcall, s, 12) exited non-zero on ‘SIP/110-00000093’ in macro ‘hangupcall’
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/110-00000093’
[Aug 18 18:18:12] VERBOSE[5070] app_mixmonitor.c: == End MixMonitor Recording SIP/110-00000093
[Aug 18 18:18:19] VERBOSE[5066] pbx.c: – Executing [h@macro-dialout-trunk:1] Macro(“SIP/101-00000091”, “hangupcall,”) in new stack
[Aug 18 18:18:19] VERBOSE[5066] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/101-00000091”, “1?noautomon”) in new stack
[Aug 18 18:18:19] VERBOSE[5066] pbx.c: – Goto (macro-hangupcall,s,3)
[Aug 18 18:18:19] VERBOSE[5066] pbx.c: – Executing [s@macro-hangupcall:3] NoOp(“SIP/101-00000091”, “TOUCH_MONITOR_OUTPUT=”) in new stack
[Aug 18 18:18:19] VERBOSE[5066] pbx.c: – Executing [s@macro-hangupcall:4] GotoIf(“SIP/101-00000091”, “1?skiprg”) in new stack
[Aug 18 18:18:19] VERBOSE[5066] pbx.c: – Goto (macro-hangupcall,s,7)
[Aug 18 18:18:19] VERBOSE[5066] pbx.c: – Executing [s@macro-hangupcall:7] GotoIf(“SIP/101-00000091”, “1?skipblkvm”) in new stack
[Aug 18 18:18:19] VERBOSE[5066] pbx.c: – Goto (macro-hangupcall,s,10)
[Aug 18 18:18:19] VERBOSE[5066] pbx.c: – Executing [s@macro-hangupcall:10] GotoIf(“SIP/101-00000091”, “1?theend”) in new stack
[Aug 18 18:18:19] VERBOSE[5066] pbx.c: – Goto (macro-hangupcall,s,12)
[Aug 18 18:18:19] VERBOSE[5066] pbx.c: – Executing [s@macro-hangupcall:12] Hangup(“SIP/101-00000091”, “”) in new stack
[Aug 18 18:18:19] VERBOSE[5066] app_macro.c: == Spawn extension (macro-hangupcall, s, 12) exited non-zero on ‘SIP/101-00000091’ in macro ‘hangupcall’
[Aug 18 18:18:19] VERBOSE[5066] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/101-00000091’ in macro ‘dialout-trunk’
[Aug 18 18:18:19] VERBOSE[5066] pbx.c: == Spawn extension (from-internal, 4276100, 5) exited non-zero on ‘SIP/101-00000091’
[Aug 18 18:18:19] VERBOSE[5067] app_mixmonitor.c: == MixMonitor close filestream
[Aug 18 18:18:19] VERBOSE[5067] app_mixmonitor.c: == End MixMonitor Recording SIP/101-00000091

check some packets captured in wireshark

1056 28.557905 192.168.0.12 192.168.0.2 ICMP 134 Destination unreachable (Port unreachable)
899 27.882355 192.168.0.2 192.168.0.11 SIP 587 Status: 486 Busy Here
893 27.864444 192.168.0.12 192.168.0.2 SIP 539 Status: 486 Busy Here
892 27.857346 192.168.0.12 192.168.0.2 SIP 479 Status: 100 Trying
831 27.591680 192.168.0.2 192.168.0.11 SIP 546 Status: 401 Unauthorized
830 27.591028 192.168.0.11 192.168.0.2 SIP/SDP 1064 Request: INVITE sip:2384375@192.168.0.2, with session description

my trunks configs:

primary trunk (192.168.0.11)

disallow=all
allow=ulaw,alaw,g711,g723,g726,g729
canreinvite=yes
context=from-trunk
dtmfmode=rfc2833
host=dynamic
incominglimit=2
nat=yes
qualify=yes
secret=7698538
type=friend
username=7698538
insecure=invite

secondary trunk (192.168.0.12)

disallow=all
allow=ulaw,alaw,g711,g723,g726,g729
canreinvite=yes
context=from-trunk
dtmfmode=rfc2833
host=dynamic
incominglimit=2
nat=yes
qualify=yes
secret=8005001
type=friend
username=8005001
fromuser=8005001
insecure=invite

I hope someone can help me, I have had this problem for more than 6 months, I have searched anywhere with no success

david55 · August 19, 2011, 6:51am

For a short form trace, sip set history output would be more useful. For a full trace, sip set debug output. There seem to be missing events in your trace.

Busy here is a very simple error. The only ambiguity is whether the HT03 is simply reporting that its PSTN line is off hook, or is actually trying to dial and getting a busy tone.

It doesn’t indicate any problem with Asterisk.

I assume that the HT03 is some sort of FXO side ATA.

system · August 19, 2011, 9:41am

Well, great work! You have helped me to improve my knowledge about this field. Thank you so much for sharing.

system · August 19, 2011, 5:11pm

thanks for answering david55

both trunks, and both HT503 allow calling ok in a independent way, the problem comes when the first one trunk (ht503#1 (192.168.0.11)) is busy, asterisk should call using the second one, (ht503#2 (192.168.0.12))

in this moment, error comes!! check this part of the capture, I think error is there, but no idea how can I solve it

[color=#FF0000][Aug 18 18:18:12] VERBOSE[5069] app_dial.c: – Called 8005001/2384375
[Aug 18 18:18:12] VERBOSE[3407] chan_sip.c: – Got SIP response 486 “Busy Here” back from 192.168.0.12
[Aug 18 18:18:12] VERBOSE[5069] app_dial.c: – SIP/8005001-00000094 is busy
[Aug 18 18:18:12] VERBOSE[5069] app_dial.c: == Everyone is busy/congested at this time (1:1/0/0)
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s@macro-dialout-trunk:20] NoOp(“SIP/110-00000093”, “Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s@macro-dialout-trunk:21] Goto(“SIP/110-00000093”, “s-BUSY,1”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Goto (macro-dialout-trunk,s-BUSY,1)
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s-BUSY@macro-dialout-trunk:1] NoOp(“SIP/110-00000093”, “Dial failed due to trunk reporting BUSY - giving up”) in new stack
[Aug 18 18:18:12] VERBOSE[5069] pbx.c: – Executing [s-BUSY@macro-dialout-trunk:2] PlayTones(“SIP/110-00000093”, “busy”) in new stack
[color=#000080][Aug 18 18:18:12] WARNING[5069] translate.c: no samples for ulawtolin[/color] [/color]

regarding missing events in my trace; I have captured a part of the log when the error comes, I turned asterisk off, and then turned it on, with the purpose to get a clean capture when the problem happens

thank you so much,

I still hope you or someone else can help me

Carlos Melo

david55 · August 19, 2011, 5:18pm

It is calling using the second one, and it is getting a busy response from that. You need to find why the second one is responding busy. Alternatively, it is not actually trying the first one at all - there isn’t enough trace to be sure, and, as you don’t have verbose logging on, one can’t see the dialplan it is trying to execute.

[Aug 18 18:18:12] VERBOSE[3407] chan_sip.c: – Got SIP response 486 “Busy Here” back from 192.168.0.1[color=#FF0000]2[/color]

As presented here so far, this is a problem with the telephone line, or the HT503, not with Asterisk.

david55 · August 19, 2011, 5:28pm

Sorry, you do have verbose on. Subject to seeing the actual complete dialplan, it appears to me that your dialplan only ever tries to make calls on the second HT503.

system · August 19, 2011, 8:06pm

david55, thanks for your advices,

now that you tell me, it seems that my problem is the dial plan, to be honest, I have not never used one, why?? because I am introducing myself into asterisk,

in my outboud routes, I set NXXXXXX , numbers that I always dial have seven digits, but about extensions.conf I have not idea regarding the way to set a dial plan.

Carlos Melo

david55 · August 19, 2011, 9:02pm

asteriskdocs.org/
…configs/extensions.conf.sample in the source code.

system · August 20, 2011, 2:27am

These questions should be asked in the GUI section, freepbx.org or the distro used.
The OP did not specified the distro, but my crystal ball says Elastix

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TWO HT503 Got SIP response 486 "Busy Here" back from"

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