Help: all circuit busy

Asterisk Asterisk Support
1

Hi all,

I am new to asterisk. I have followed all the instruction to install asterisk and freepbx. I then setup a sip provider and an extension for my local phone. It looks like the sip registration went well. However every time that I made an outgoing call, I got the error message ‘all circuit busy’. Would somebody able to help me out.

btw, when I pick up the phone to dial, I didn’t hear any dial tone either.

thank you very much,

Here is my call log
localhostCLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [5083409818@from-internal:1] Macro(“SIP/100-00000004”, “user-callerid,LIMIT,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/100-00000004”, “AMPUSER=100”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/100-00000004”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/100-00000004”, “1?Set(REALCALLERIDNUM=100)”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/100-00000004”, “AMPUSER=100”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/100-00000004”, “AMPUSERCIDNAME=100”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/100-00000004”, “0?report”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/100-00000004”, “AMPUSERCID=100”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/100-00000004”, “CALLERID(all)=“100” <100>”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“SIP/100-00000004”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:10] ExecIf(“SIP/100-00000004”, “1?Set(GROUP(concurrency_limit)=100)”) in new stack
– Executing [s@macro-user-callerid:11] GosubIf(“SIP/100-00000004”, “0?sub-ccss,s,1(from-internal,5083409818)”) in new stack
– Executing [s@macro-user-callerid:12] GotoIf(“SIP/100-00000004”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,25)
– Executing [s@macro-user-callerid:25] Set(“SIP/100-00000004”, “CALLERID(number)=100”) in new stack
– Executing [s@macro-user-callerid:26] Set(“SIP/100-00000004”, “CALLERID(name)=100”) in new stack
– Executing [s@macro-user-callerid:27] Set(“SIP/100-00000004”, “CHANNEL(language)=en”) in new stack
– Executing [5083409818@from-internal:2] Set(“SIP/100-00000004”, “MOHCLASS=default”) in new stack
– Executing [5083409818@from-internal:3] Set(“SIP/100-00000004”, “_NODEST=”) in new stack
– Executing [5083409818@from-internal:4] Macro(“SIP/100-00000004”, “record-enable,100,OUT,”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/100-00000004”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] ExecIf(“SIP/100-00000004”, “0?MacroExit()”) in new stack
– Executing [s@macro-record-enable:5] GotoIf(“SIP/100-00000004”, “0?Group:OUT”) in new stack
– Goto (macro-record-enable,s,14)
– Executing [s@macro-record-enable:14] GotoIf(“SIP/100-00000004”, “0?IN”) in new stack
– Executing [s@macro-record-enable:15] ExecIf(“SIP/100-00000004”, “1?MacroExit()”) in new stack
– Executing [5083409818@from-internal:5] Macro(“SIP/100-00000004”, “dialout-trunk,4,5083409818,”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/100-00000004”, “DIAL_TRUNK=4”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/100-00000004”, “0?sub-pincheck,s,1”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/100-00000004”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/100-00000004”, “DIAL_NUMBER=5083409818”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/100-00000004”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/100-00000004”, “OUTBOUND_GROUP=OUT_4”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/100-00000004”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/100-00000004”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/100-00000004”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/100-00000004”, “outbound-callerid,4”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/100-00000004”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/100-00000004”, “0?Set(REALCALLERIDNUM=100)”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/100-00000004”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/100-00000004”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/100-00000004”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/100-00000004”, “TRUNKOUTCID=5087099119”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/100-00000004”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] ExecIf(“SIP/100-00000004”, “1?Set(CALLERID(all)=5087099119)”) in new stack
– Executing [s@macro-outbound-callerid:13] ExecIf(“SIP/100-00000004”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:14] ExecIf(“SIP/100-00000004”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:15] ExecIf(“SIP/100-00000004”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [s@macro-dialout-trunk:12] GosubIf(“SIP/100-00000004”, “1?sub-flp-4,s,1”) in new stack
– Executing [s@sub-flp-4:1] ExecIf(“SIP/100-00000004”, “1?Return()”) in new stack
– Executing [s@macro-dialout-trunk:13] Set(“SIP/100-00000004”, “OUTNUM=5083409818”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/100-00000004”, “custom=SIP/sipsorcery”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/100-00000004”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))”) in new stack
– Executing [s@macro-dialout-trunk:16] ExecIf(“SIP/100-00000004”, “0?Set(DIAL_TRUNK_OPTIONS=M(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:17] Macro(“SIP/100-00000004”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/100-00000004”, “”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/100-00000004”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:19] GotoIf(“SIP/100-00000004”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:20] Dial(“SIP/100-00000004”, “SIP/sipsorcery/5083409818,300,”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/sipsorcery/5083409818
– SIP/sipsorcery-00000005 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [s@macro-dialout-trunk:21] NoOp(“SIP/100-00000004”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 1”) in new stack
– Executing [s@macro-dialout-trunk:22] Goto(“SIP/100-00000004”, “s-CONGESTION,1”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,1)
– Executing [s-CONGESTION@macro-dialout-trunk:1] Set(“SIP/100-00000004”, “RC=1”) in new stack
– Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(“SIP/100-00000004”, “1,1”) in new stack
– Goto (macro-dialout-trunk,1,1)
– Executing [1@macro-dialout-trunk:1] Goto(“SIP/100-00000004”, “continue,1”) in new stack
– Goto (macro-dialout-trunk,continue,1)
– Executing [continue@macro-dialout-trunk:1] GotoIf(“SIP/100-00000004”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,continue,3)
– Executing [continue@macro-dialout-trunk:3] NoOp(“SIP/100-00000004”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 1 - failing through to other trunks”) in new stack
– Executing [continue@macro-dialout-trunk:4] Set(“SIP/100-00000004”, “CALLERID(number)=100”) in new stack
– Executing [5083409818@from-internal:6] Macro(“SIP/100-00000004”, “outisbusy,”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“SIP/100-00000004”, “”) in new stack
– Executing [s@macro-outisbusy:2] GotoIf(“SIP/100-00000004”, “0?emergency,1”) in new stack
– Executing [s@macro-outisbusy:3] GotoIf(“SIP/100-00000004”, “0?intracompany,1”) in new stack
– Executing [s@macro-outisbusy:4] Playback(“SIP/100-00000004”, “all-circuits-busy-now&pls-try-call-later, noanswer”) in new stack
[Sep 7 08:19:37] WARNING[934]: file.c:653 ast_openstream_full: File all-circuits-busy-now does not exist in any format
[Sep 7 08:19:37] WARNING[934]: file.c:959 ast_streamfile: Unable to open all-circuits-busy-now (format 0x4 (ulaw)): No such file or directory
[Sep 7 08:19:37] WARNING[934]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/100-00000004 for all-circuits-busy-now&pls-try-call-later, noanswer
[Sep 7 08:19:37] WARNING[934]: file.c:653 ast_openstream_full: File pls-try-call-later does not exist in any format
[Sep 7 08:19:37] WARNING[934]: file.c:959 ast_streamfile: Unable to open pls-try-call-later (format 0x4 (ulaw)): No such file or directory
[Sep 7 08:19:37] WARNING[934]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/100-00000004 for all-circuits-busy-now&pls-try-call-later, noanswer
– Executing [s@macro-outisbusy:5] Congestion(“SIP/100-00000004”, “20”) in new stack
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on ‘SIP/100-00000004’ in macro ‘outisbusy’
== Spawn extension (from-internal, 5083409818, 6) exited non-zero on ‘SIP/100-00000004’
– Executing [h@from-internal:1] Hangup(“SIP/100-00000004”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/100-00000004’
> doing dnsmgr_lookup for ‘sipsorcery.com’
> ast_get_srv: SRV lookup for ‘_sip._udp.sipsorcery.com’ mapped to host sip2.sipsorcery.com, port 5060
[Sep 7 08:19:43] WARNING[526]: chan_sip.c:3620 retrans_pkt: Retransmission timeout reached on transmission 762931139676@192.168.1.137 for seqno 2 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 6400ms with no response
localhostCLI>
localhost*CLI>

2

Hi redmapleleaf!

I think you need to give us the content of your sip.conf file!
Make sure that you remove all of you passwords in it before you post it here!

What reply do you get if you in the asterisk console type:

What reply do you get with:

and see that your registration with your sip provider is OK.

Virtually yours // Nypon

3

Which version of Asterisk is this? Hangupcause 1 is number does not exist, not busy. It should not be summarising that as CONGESTION.

4

hi,

thank you for the reply.

I used the latest stable asterisk 1.8.6 .

when I type ‘sip show registry’, I got

localhostCLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
sipsorcery.com:5060 N xxxx 105 Registered Wed, 07 Sep 2011 11:11:45
1 SIP registrations.
localhostCLI>

localhostCLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
100/100 192.168.1.137 D A 38167 OK (178 ms)
sipsorcery/xxxx 69.59.142.213 5060 OK (104 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
localhostCLI>

The sip_additional.conf has this content

;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;

[100]
deny=0.0.0.0/0.0.0.0
secret=xxxxxx
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
encryption=no
callgroup=
pickupgroup=
dial=SIP/100
mailbox=100@device
permit=0.0.0.0/0.0.0.0
callerid=device <100>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

[sipsorcery]
disallow=all
type=peer
host=sipsorcery.com
outboundproxy=sipsorcery.com
outboundproxy=sipsorcery.com
insecure=invite
qualify=yes
dtmfmode=rfc2833
username=xxxx
defaultuser=xxxx
fromuser=xxxx
secret=xxxx
context=sipsorcery_in
allow=ulaw

5

it is working better now that I follow the instruction from freepbx.org/forum/freepbx/in … stallation

and added these steps:

contrib/scripts/get_mp3_source.sh
Run configure
./configure
Start menu based configuration
make menuconfig
select app_mysql, app_saycountpl, cdr_mysql, format_mp3, res_config_mysql
go to Extras Sound Packages
Select EXTRAS-SOUNDS-EN-GSM
Save & Exit
Run make
make
Then run the installer
make install

I am able to hear the dial tone when making a call. However, I get a new database error:

== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/100-00000000’
[Sep 7 11:52:03] ERROR[17599]: cdr_mysql.c:203 mysql_log: Cannot connect to database server localhost: (1049) Unknown database ‘asteriskcdrdb’
[Sep 7 11:52:09] WARNING[17536]: chan_sip.c:3620 retrans_pkt: Retransmission timeout reached on transmission 164291733937@192.168.1.137 for seqno 2 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 6400ms with no response
localhost*CLI>

It seems like this database was never created and I didn’t see any installation instruction mentions about it…

6

ok, I have been able to make significant progress. I am able to receive incoming call now. To solve the previous problem of missing asteriskcdrdb by executing the command

mysqladmin -u root -p create asteriskcdrdb

to create the required database.

I also added an extension definition in exentions_custom.conf file

[sipsorcery_in]
exten => 100,1,Dial(SIP/100,25,t)

This allows incoming call to be routed to extension 100 and is picked up by my sipphone. Everything working great.

The problem that I have now is on the outgoing call. Every time I dial out from my sipphone at extension 100, I got the recording “All circuits are busy now, please try your call later”. I am not sure what problem this is yet. Here is my calling log:

localhostCLI>
localhostCLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [5083409828@from-internal:1] Macro(“SIP/100-00000005”, “user-callerid,LIMIT,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/100-00000005”, “AMPUSER=100”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/100-00000005”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/100-00000005”, “1?Set(REALCALLERIDNUM=100)”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/100-00000005”, “AMPUSER=100”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/100-00000005”, “AMPUSERCIDNAME=100”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/100-00000005”, “0?report”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/100-00000005”, “AMPUSERCID=100”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/100-00000005”, “CALLERID(all)=“100” <100>”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“SIP/100-00000005”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:10] ExecIf(“SIP/100-00000005”, “1?Set(GROUP(concurrency_limit)=100)”) in new stack
– Executing [s@macro-user-callerid:11] GosubIf(“SIP/100-00000005”, “0?sub-ccss,s,1(from-internal,5083409828)”) in new stack
– Executing [s@macro-user-callerid:12] GotoIf(“SIP/100-00000005”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,25)
– Executing [s@macro-user-callerid:25] Set(“SIP/100-00000005”, “CALLERID(number)=100”) in new stack
– Executing [s@macro-user-callerid:26] Set(“SIP/100-00000005”, “CALLERID(name)=100”) in new stack
– Executing [s@macro-user-callerid:27] Set(“SIP/100-00000005”, “CHANNEL(language)=en”) in new stack
– Executing [5083409828@from-internal:2] Set(“SIP/100-00000005”, “MOHCLASS=default”) in new stack
– Executing [5083409828@from-internal:3] Set(“SIP/100-00000005”, “_NODEST=”) in new stack
– Executing [5083409828@from-internal:4] Macro(“SIP/100-00000005”, “record-enable,100,OUT,”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/100-00000005”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] ExecIf(“SIP/100-00000005”, “0?MacroExit()”) in new stack
– Executing [s@macro-record-enable:5] GotoIf(“SIP/100-00000005”, “0?Group:OUT”) in new stack
– Goto (macro-record-enable,s,14)
– Executing [s@macro-record-enable:14] GotoIf(“SIP/100-00000005”, “0?IN”) in new stack
– Executing [s@macro-record-enable:15] ExecIf(“SIP/100-00000005”, “1?MacroExit()”) in new stack
– Executing [5083409828@from-internal:5] Macro(“SIP/100-00000005”, “dialout-trunk,4,5083409828,”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/100-00000005”, “DIAL_TRUNK=4”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/100-00000005”, “0?sub-pincheck,s,1”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/100-00000005”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/100-00000005”, “DIAL_NUMBER=5083409828”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/100-00000005”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/100-00000005”, “OUTBOUND_GROUP=OUT_4”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/100-00000005”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/100-00000005”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/100-00000005”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/100-00000005”, “outbound-callerid,4”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/100-00000005”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/100-00000005”, “0?Set(REALCALLERIDNUM=100)”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/100-00000005”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/100-00000005”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/100-00000005”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/100-00000005”, “TRUNKOUTCID=5087099119”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/100-00000005”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] ExecIf(“SIP/100-00000005”, “1?Set(CALLERID(all)=5087099119)”) in new stack
– Executing [s@macro-outbound-callerid:13] ExecIf(“SIP/100-00000005”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:14] ExecIf(“SIP/100-00000005”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:15] ExecIf(“SIP/100-00000005”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [s@macro-dialout-trunk:12] GosubIf(“SIP/100-00000005”, “1?sub-flp-4,s,1”) in new stack
– Executing [s@sub-flp-4:1] ExecIf(“SIP/100-00000005”, “1?Return()”) in new stack
– Executing [s@macro-dialout-trunk:13] Set(“SIP/100-00000005”, “OUTNUM=5083409828”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/100-00000005”, “custom=SIP/sipsorcery”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/100-00000005”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))”) in new stack
– Executing [s@macro-dialout-trunk:16] ExecIf(“SIP/100-00000005”, “0?Set(DIAL_TRUNK_OPTIONS=M(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:17] Macro(“SIP/100-00000005”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/100-00000005”, “”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/100-00000005”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:19] GotoIf(“SIP/100-00000005”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:20] Dial(“SIP/100-00000005”, “SIP/sipsorcery/5083409828,300,”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/sipsorcery/5083409828
– SIP/sipsorcery-00000006 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [s@macro-dialout-trunk:21] NoOp(“SIP/100-00000005”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 1”) in new stack
– Executing [s@macro-dialout-trunk:22] Goto(“SIP/100-00000005”, “s-CONGESTION,1”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,1)
– Executing [s-CONGESTION@macro-dialout-trunk:1] Set(“SIP/100-00000005”, “RC=1”) in new stack
– Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(“SIP/100-00000005”, “1,1”) in new stack
– Goto (macro-dialout-trunk,1,1)
– Executing [1@macro-dialout-trunk:1] Goto(“SIP/100-00000005”, “continue,1”) in new stack
– Goto (macro-dialout-trunk,continue,1)
– Executing [continue@macro-dialout-trunk:1] GotoIf(“SIP/100-00000005”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,continue,3)
– Executing [continue@macro-dialout-trunk:3] NoOp(“SIP/100-00000005”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 1 - failing through to other trunks”) in new stack
– Executing [continue@macro-dialout-trunk:4] Set(“SIP/100-00000005”, “CALLERID(number)=100”) in new stack
– Executing [5083409828@from-internal:6] Macro(“SIP/100-00000005”, “outisbusy,”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“SIP/100-00000005”, “”) in new stack
– Executing [s@macro-outisbusy:2] GotoIf(“SIP/100-00000005”, “0?emergency,1”) in new stack
– Executing [s@macro-outisbusy:3] GotoIf(“SIP/100-00000005”, “0?intracompany,1”) in new stack
– Executing [s@macro-outisbusy:4] Playback(“SIP/100-00000005”, “all-circuits-busy-now&pls-try-call-later, noanswer”) in new stack
– <SIP/100-00000005> Playing ‘all-circuits-busy-now.gsm’ (language ‘en’)
– <SIP/100-00000005> Playing ‘pls-try-call-later.gsm’ (language ‘en’)
– Executing [s@macro-outisbusy:5] Congestion(“SIP/100-00000005”, “20”) in new stack
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on ‘SIP/100-00000005’ in macro ‘outisbusy’
== Spawn extension (from-internal, 5083409828, 6) exited non-zero on ‘SIP/100-00000005’
– Executing [h@from-internal:1] Hangup(“SIP/100-00000005”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000005’
localhost*CLI>

7

okay, I finally fix my outgoing circuit busy problem, ending my pbx configuration misery.

I did it by changing my trunk definition from

[sipsorcery]
disallow=all
type=peer
host=sipsorcery.com
outboundproxy=sipsorcery.com
insecure=invite
qualify=yes
dtmfmode=rfc2833
username=xxxx
defaultuser=xxxx
fromuser=xxxx
secret=xxxx
context=sipsorcery_in
allow=ulaw

to

[sipsorcery]
context=sipsorcery_in
fromdomain=sipsorcery.com
fromuser=xxxx
host=sipsorcery.com
insecure=port,invite
secret=xxxx
type=peer
defaultuser=xxxx

I hope that the information here could help some other newbies out there. If you happen to run into asterisk configuration problem and are reading this message, then hey I share your pain…

8

Have you tried it with insecure=invite unchanged. insecure=port,invite is often recommended when it isn’t needed.

9

redmapleleaf

I am indebted to you. I had been tearing my hair off for the last few days unable to make any outgoing calls, but your solution WORKED!!! :smiley:

[quote=“redmapleleaf”]okay, I finally fix my outgoing circuit busy problem, ending my pbx configuration misery.

I did it by changing my trunk definition from

[sipsorcery]
disallow=all
type=peer
host=sipsorcery.com
outboundproxy=sipsorcery.com
insecure=invite
qualify=yes
dtmfmode=rfc2833
username=xxxx
defaultuser=xxxx
fromuser=xxxx
secret=xxxx
context=sipsorcery_in
allow=ulaw

to

[sipsorcery]
context=sipsorcery_in
fromdomain=sipsorcery.com
fromuser=xxxx
host=sipsorcery.com
insecure=port,invite
secret=xxxx
type=peer
defaultuser=xxxx

I hope that the information here could help some other newbies out there. If you happen to run into asterisk configuration problem and are reading this message, then hey I share your pain…[/quote]