Cannot dial internal extension "all circuits are busy"

Dear all,

I inherited our SIP server and I am in no way a VoIP expert! I am having a problem when dialling an internal number from another extension. The number used to dial fine but it now stops and says “I’m sorry but all circuits are busy at the moment, please try your call again later”. After it has said this, I get some beeps on the line and the phone says “Address incomplete”.

I have pasted what happens in the asterisk CLI when I dial the extension (283) from my extension (668). I’m not sure where to start looking but it seems to only happen with extensions that begin with 2xx if that helps…?!

Thanks for anyone’s input.

Jon.

<— SIP read from 10.92.20.4:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.92.20.4:5060;branch=z9hG4bK-9b85f86e
From: “Jon Smith” sip:[email protected];tag=bfc83c6d5e0c37ceo0
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: “Jon Smith” sip:[email protected]:5060
Expires: 240
User-Agent: Cisco/SPA504G-7.4.3a
Content-Length: 391
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 155353 155353 IN IP4 10.92.20.4
s=-
c=IN IP4 10.92.20.4
t=0 0
m=audio 16402 RTP/AVP 0 2 8 9 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (14 headers 18 lines) —
Sending to 10.92.20.4 : 5060 (NAT)
Using INVITE request as basis request - [email protected]
sip*CLI>
<— Reliably Transmitting (NAT) to 10.92.20.4:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.92.20.4:5060;branch=z9hG4bK-9b85f86e;received=10.92.20.4
From: “Jon Smith” sip:[email protected];tag=bfc83c6d5e0c37ceo0
To: sip:[email protected];tag=as7458bf6b
Call-ID: [email protected]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4d3842ec"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
Found user '668’
sip*CLI>
<— SIP read from 10.92.20.4:5060 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.92.20.4:5060;branch=z9hG4bK-9b85f86e
From: “Jon Smith” sip:[email protected];tag=bfc83c6d5e0c37ceo0
To: sip:[email protected];tag=as7458bf6b
Call-ID: [email protected]
CSeq: 101 ACK
Max-Forwards: 70
Contact: “Jon Smith” sip:[email protected]:5060
User-Agent: Cisco/SPA504G-7.4.3a
Content-Length: 0

<------------->
— (10 headers 0 lines) —
sip*CLI>
<— SIP read from 10.92.20.4:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.92.20.4:5060;branch=z9hG4bK-cf24507f
From: “Jon Smith” sip:[email protected];tag=bfc83c6d5e0c37ceo0
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username=“668”,realm=“asterisk”,nonce=“4d3842ec”,uri="sip:[email protected]",algorithm=MD5,response="6d9068bf8f02e65167f1d48ceb801fc9"
Contact: “Jon Smith” sip:[email protected]:5060
Expires: 240
User-Agent: Cisco/SPA504G-7.4.3a
Content-Length: 391
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 155353 155353 IN IP4 10.92.20.4
s=-
c=IN IP4 10.92.20.4
t=0 0
m=audio 16402 RTP/AVP 0 2 8 9 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (15 headers 18 lines) —
Sending to 10.92.20.4 : 5060 (NAT)
Using INVITE request as basis request - [email protected]
Found user '668’
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 10.92.20.4:16402
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format G729a for ID 18
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.92.20.4:16402
Looking for 283 in ncos-6 (domain 10.10.1.1)
list_route: hop: sip:[email protected]:5060
sip*CLI>
<— Transmitting (NAT) to 10.92.20.4:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.92.20.4:5060;branch=z9hG4bK-cf24507f;received=10.92.20.4
From: “Jon Smith” sip:[email protected];tag=bfc83c6d5e0c37ceo0
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
– Executing [[email protected]:1] Macro(“SIP/668-b72042f8”, “user-callerid”) in new stack
– Executing [[email protected]:1] NoOp(“SIP/668-b72042f8”, “user-callerid: device 668”) in new stack
– Executing [[email protected]:2] Set(“SIP/668-b72042f8”, “AMPUSER=668”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/668-b72042f8”, “0?report”) in new stack
– Executing [[email protected]:4] ExecIf(“SIP/668-b72042f8”, “1|Set|REALCALLERIDNUM=668”) in new stack
– Executing [[email protected]:5] NoOp(“SIP/668-b72042f8”, “REALCALLERIDNUM is 668”) in new stack
– Executing [[email protected]:6] Set(“SIP/668-b72042f8”, “AMPUSER=668”) in new stack
– Executing [[email protected]:7] Set(“SIP/668-b72042f8”, "AMPUSERCIDNAME=Jon Smith ") in new stack
– Executing [[email protected]:8] GotoIf(“SIP/668-b72042f8”, “0?report”) in new stack
– Executing [[email protected]:9] Set(“SIP/668-b72042f8”, “AMPUSERCID=557”) in new stack
– Executing [[email protected]:10] Set(“SIP/668-b72042f8”, "CALLERID(all)=“Jon Smith " <557>”) in new stack
– Executing [[email protected]:11] Set(“SIP/668-b72042f8”, “REALCALLERIDNUM=668”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/668-b72042f8”, “0|Set|CHANNEL(language)=”) in new stack
– Executing [[email protected]:13] NoOp(“SIP/668-b72042f8”, "TTL: ARG1: ") in new stack
– Executing [[email protected]:14] GotoIf(“SIP/668-b72042f8”, “0?continue”) in new stack
– Executing [[email protected]:15] Set(“SIP/668-b72042f8”, “__TTL=64”) in new stack
– Executing [[email protected]:16] GotoIf(“SIP/668-b72042f8”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,23)
– Executing [[email protected]:23] NoOp(“SIP/668-b72042f8”, “Using CallerID “Jon Smith” <557>”) in new stack
– Executing [[email protected]:2] Dial(“SIP/668-b72042f8”, “ZAP/g0/01684544283”) in new stack
– Requested transfer capability: 0x00 - SPEECH
– Called g0/01684544283
Audio is at 10.10.1.1 port 15126
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
sip*CLI>
<— Transmitting (NAT) to 10.92.20.4:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.92.20.4:5060;branch=z9hG4bK-cf24507f;received=10.92.20.4
From: “Jon Smith” sip:[email protected];tag=bfc83c6d5e0c37ceo0
To: sip:[email protected];tag=as7f121c7c
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 28760 28760 IN IP4 10.10.1.1
s=session
c=IN IP4 10.10.1.1
t=0 0
m=audio 15126 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– Zap/3-1 is proceeding passing it to SIP/668-b72042f8
– Got SIP response 400 “Bad Request” back from 10.90.20.7
– Got SIP response 400 “Bad Request” back from 10.3.20.13
– Channel 0/3, span 1 got hangup request, cause 1
– Hungup ‘Zap/3-1’
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [[email protected]:3] Macro(“SIP/668-b72042f8”, “record-enable|668|OUT|”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/668-b72042f8”, “0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/668-b72042f8”, “recordingcheck|20111027-095529|1319705727.47429”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20111027-095529|1319705727.47429: Outbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:5] NoOp(“SIP/668-b72042f8”, “No recording needed”) in new stack
– Executing [[email protected]:4] Macro(“SIP/668-b72042f8”, “dialout-trunk|1|283||”) in new stack
– Executing [[email protected]:1] Set(“SIP/668-b72042f8”, “DIAL_TRUNK=1”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/668-b72042f8”, “0|Authenticate|”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/668-b72042f8”, “0?disabletrunk|1”) in new stack
– Executing [[email protected]:4] Set(“SIP/668-b72042f8”, “DIAL_NUMBER=283”) in new stack
– Executing [[email protected]:5] Set(“SIP/668-b72042f8”, “DIAL_TRUNK_OPTIONS=trTwW”) in new stack
– Executing [[email protected]:6] Set(“SIP/668-b72042f8”, “GROUP()=OUT_1”) in new stack
– Executing [[email protected]:7] GotoIf(“SIP/668-b72042f8”, “0?nomax”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/668-b72042f8”, “0?chanfull”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/668-b72042f8”, “0?skipoutcid”) in new stack
– Executing [[email protected]:10] Set(“SIP/668-b72042f8”, “DIAL_TRUNK_OPTIONS=tTwW”) in new stack
– Executing [[email protected]:11] Macro(“SIP/668-b72042f8”, “outbound-callerid|1”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/668-b72042f8”, “1?start”) in new stack
– Goto (macro-outbound-callerid,s,3)
– Executing [[email protected]:3] NoOp(“SIP/668-b72042f8”, “REALCALLERIDNUM is 668”) in new stack
– Executing [[email protected]:4] GotoIf(“SIP/668-b72042f8”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,9)
– Executing [[email protected]:9] Set(“SIP/668-b72042f8”, “USEROUTCID=”) in new stack
– Executing [[email protected]:10] Set(“SIP/668-b72042f8”, “EMERGENCYCID=”) in new stack
– Executing [[email protected]:11] Set(“SIP/668-b72042f8”, “TRUNKOUTCID=”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/668-b72042f8”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,16)
– Executing [[email protected]:16] GotoIf(“SIP/668-b72042f8”, “1?usercid”) in new stack
– Goto (macro-outbound-callerid,s,18)
– Executing [[email protected]:18] GotoIf(“SIP/668-b72042f8”, “1?report”) in new stack
– Goto (macro-outbound-callerid,s,22)
– Executing [[email protected]:22] NoOp(“SIP/668-b72042f8”, "CallerID set to “Jon Smith " <557>”) in new stack
– Executing [[email protected]:12] AGI(“SIP/668-b72042f8”, “fixlocalprefix”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern 9|.
> fixlocalprefix: Using pattern .|
> fixlocalprefix: Using pattern .
== fixlocalprefix: Dialpattern . matched. 283 -> 283
– AGI Script fixlocalprefix completed, returning 0
– Executing [[email protected]:13] Set(“SIP/668-b72042f8”, “OUTNUM=283”) in new stack
– Executing [[email protected]:14] Set(“SIP/668-b72042f8”, “custom=ZAP/g0”) in new stack
– Executing [[email protected]:15] GotoIf(“SIP/668-b72042f8”, “1?gocall”) in new stack
– Goto (macro-dialout-trunk,s,17)
– Executing [[email protected]:17] Macro(“SIP/668-b72042f8”, “dialout-trunk-predial-hook|”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/668-b72042f8”, “0?bypass|1”) in new stack
– Executing [[email protected]:19] GotoIf(“SIP/668-b72042f8”, “0?customtrunk”) in new stack
– Executing [[email protected]:20] Dial(“SIP/668-b72042f8”, “ZAP/g0/283|300|tTwW”) in new stack
– Requested transfer capability: 0x00 - SPEECH
– Called g0/283
– Zap/3-1 is proceeding passing it to SIP/668-b72042f8
– Got SIP response 400 “Bad Request” back from 10.90.20.15
– Channel 0/3, span 1 got hangup request, cause 28
– Hungup ‘Zap/3-1’
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [[email protected]:21] Goto(“SIP/668-b72042f8”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing [[email protected]:1] GotoIf(“SIP/668-b72042f8”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
– Executing [[email protected]:3] NoOp(“SIP/668-b72042f8”, “TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks”) in new stack
– Executing [[email protected]:5] Macro(“SIP/668-b72042f8”, “outisbusy|”) in new stack
– Executing [[email protected]:1] Playback(“SIP/668-b72042f8”, “all-circuits-busy-now|noanswer”) in new stack
– <SIP/668-b72042f8> Playing ‘all-circuits-busy-now’ (language ‘en’)
– Executing [[email protected]:2] Playback(“SIP/668-b72042f8”, “pls-try-call-later|noanswer”) in new stack
– <SIP/668-b72042f8> Playing ‘pls-try-call-later’ (language ‘en’)
sip*CLI>