Outbound calls through Sip error

General Help
1

Hi,

Whenever I try calling using SIP port I am getting this error.

Called SIP/pfingo/63536682
[2013-02-01 13:29:07] VERBOSE[5703] app_dial.c: – SIP/pfingo-00000007 is circuit-busy
[2013-02-01 13:29:07] VERBOSE[5703] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)
[2013-02-01 13:29:07] VERBOSE[5703] pbx.c: – Executing [[email protected]:23] NoOp(“SIP/134-00000006”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 0”) in new stack
[2013-02-01 13:29:07] VERBOSE[5703] pbx.c: – Executing [[email protected]:24] Goto(“SIP/134-00000006”, “s-CONGESTION,1”) in new stack
[2013-02-01 13:29:07] VERBOSE[5703] pbx.c: – Goto (macro-dialout-trunk,s-CONGESTION,1)
[2013-02-01 13:29:07] VERBOSE[5703] pbx.c: – Executing [[email protected]:1] Set(“SIP/134-00000006”, “RC=0”) in new stack
[2013-02-01 13:29:07] VERBOSE[5703] pbx.c: – Executing [[email protected]:2] Goto(“SIP/134-00000006”, “0,1”) in new stack
[2013-02-01 13:29:07] VERBOSE[5703] pbx.c: – Goto (macro-dialout-trunk,0,1)
[2013-02-01 13:29:07] VERBOSE[5703] pbx.c: – Executing [[email protected]:1] Goto(“SIP/134-00000006”, “continue,1”) in new stack
[2013-02-01 13:29:07] VERBOSE[5703] pbx.c: – Goto (macro-dialout-trunk,continue,1)
[2013-02-01 13:29:07] VERBOSE[5703] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/134-00000006”, “1?noreport”) in new stack
[2013-02-01 13:29:07] VERBOSE[5703] pbx.c: – Goto (macro-dialout-trunk,continue,3)
[2013-02-01 13:29:07] VERBOSE[5703] pbx.c: – Executing [[email protected]:3] NoOp(“SIP/134-00000006”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 0 - failing through to other trunks”) in new stack
[2013-02-01 13:29:07] VERBOSE[5703] pbx.c: – Executing [[email protected]:4] Set(“SIP/134-00000006”, “CALLERID(number)=134”) in new stack
[2013-02-01 13:29:07] VERBOSE[5703] pbx.c: – Executing [[email protected]:6] Macro(“SIP/134-00000006”, “dialout-trunk,1,63536682,”) in new stack
[2013-02-01 13:29:07] WARNING[5162] chan_sip.c: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

Any advice ?

2

check your firewall has ports 5060 and 10000-20000 open

check your sip trunk settings are correct.

check you have correctly set up an outbound route.

3