Hi,
We’ve had our Asterisk PBX (v1.4.21.2) running since last December, and it’s been working fine. However, last Thursday it suddenly stopped making outgoing calls, we just get a dead tone.
We can make internal calls, and we can receive external calls, we just can’t dial external calls. Nothing changed on the Asterisk box when it stopped working, it’s been stable, and running OK for months.
We can use a softphone like ZoIPer to connect directly to our SIP trunk, and we can make outgoing calls, so there seems to be some issue between our PBX and the SIP trunk, but I don’t know what!
I’ve had our SIP trunk supplier remote into the PBX, and have a look, and they can’t see anything wrong with the trunk settings!
The Asterisk debug log suggests the SIP trunk is busy, but as I said, we can use a softphone to dial out on the SIP trunk directly.
Can anyone think what might be wrong?
Thanks
Ben
<------------->
— (8 headers 0 lines) —
– Got SIP response 603 “Declined” back from 216.xx.xx.22
Transmitting (NAT) to 216.xx.xx.22:5060:
ACK sip:0845XXXXXX@contact-pro.co.uk SIP/2.0
Via: SIP/2.0/UDP 216.xx.xx.82:5060;branch=z9hG4bK118f6f74;rport
From: “LiteComSemi” sip:08455215050@216.xx.xx.82;tag=as204baa2c
To: sip:0845XXXXXX@contact-pro.co.uk;tag=f7d31c13e7892d3a1a66f89c80a60915.567 0
Contact: sip:08455215050@216.xx.xx.82
Call-ID: 225d6cbd3d8446a859d818df37953521@216.xx.xx.82
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
-- SIP/Contact-pro-085854c0[b] is busy[/b]
== Everyone is busy/congested at this time (1:1/0/0)
– Executing [s@macro-dialout-trunk:20] NoOp(“SIP/103-08548ce8”, “Dial faile d for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 21”) in new stack
– Executing [s@macro-dialout-trunk:21] Goto(“SIP/103-08548ce8”, “s-BUSY|1”) in new stack
– Goto (macro-dialout-trunk,s-BUSY,1)
– Executing [s-BUSY@macro-dialout-trunk:1] NoOp(“SIP/103-08548ce8”, “Dial f ailed due to trunk reporting BUSY - giving up”) in new stack
– Executing [s-BUSY@macro-dialout-trunk:2] PlayTones(“SIP/103-08548ce8”, “b usy”) in new stack
Audio is at 216.xx.xx.82 port 17230
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
pbx*CLI>
<— Transmitting (NAT) to 76.xx.xx.6:1025 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 76.xx.xx.6:1025;branch=z9hG4bK7abf9ec38413c5df1.ff5fe84a8e81c1 27f;received=76.xx.xx.6
From: “Ben” sip:103@216.xx.xx.82:5060;tag=a684eb1a81
To: “90845XXXXXX” sip:90845XXXXXX@216.xx.xx.82:5060;tag=as09eac3eb
Call-ID: ea2d1c8e24bddc7a
CSeq: 4627 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:90845XXXXXX@216.xx.xx.82
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 2816 2816 IN IP4 216.xx.xx.82
s=session
c=IN IP4 216.xx.xx.82
t=0 0
m=audio 17230 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
<------------>
– Executing [s-BUSY@macro-dialout-trunk:3] Busy(“SIP/103-08548ce8”, “20”) i n new stack
pbx*CLI>
<— Transmitting (NAT) to 76.xx.xx.6:1025 —>
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 76.xx.xx.6:1025;branch=z9hG4bK7abf9ec38413c5df1.ff5fe84a8e81c1 27f;received=76.xx.xx.6
From: “Ben” sip:103@216.xx.xx.82:5060;tag=a684eb1a81
To: “90845XXXXXX” sip:90845XXXXXX@216.xx.xx.82:5060;tag=as09eac3eb
Call-ID: ea2d1c8e24bddc7a
CSeq: 4627 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
<------------>
== Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on ‘SIP/10 3-08548ce8’ in macro ‘dialout-trunk’
== Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on ‘SIP/10 3-08548ce8’
– Executing [h@macro-dialout-trunk:1] Macro(“SIP/103-08548ce8”, “hangupcall |”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/103-08548ce8”, “1?skiprg”) i n new stack
– Goto (macro-hangupcall,s,4)
– Executing [s@macro-hangupcall:4] GotoIf(“SIP/103-08548ce8”, “1?skipblkvm” ) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] GotoIf(“SIP/103-08548ce8”, “1?theend”) i n new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] Hangup(“SIP/103-08548ce8”, “”) in new st ack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/103-08548c e8’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/103-08548c e8’
Really destroying SIP dialog '225d6cbd3d8446a859d818df37953521@216.xx.xx.82’ Met hod: INVITE
Really destroying SIP dialog ‘ea2d1c8e24bddc7a’ Method: INVITE
Retransmitting #2 (NAT) to 76.xx.xx.6:49904:
OPTIONS sip:104@76.xx.xx.6:49904;rinstance=aec6ef727a91733d SIP/2.0
Via: SIP/2.0/UDP 216.xx.xx.82:5060;branch=z9hG4bK266b980a;rport
From: “Unknown” sip:Unknown@216.xx.xx.82;tag=as09352ae6
To: sip:104@76.xx.xx.6:49904;rinstance=aec6ef727a91733d
Contact: sip:Unknown@216.xx.xx.82
Call-ID: 3090a11a4da324a00a8fec6b12e3403f@216.xx.xx.82
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 21 Sep 2010 07:34:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<— SIP read from 76.xx.xx.6:1025 —>
ACK sip:90845XXXXXX@216.xx.xx.82:5060 SIP/2.0
Via: SIP/2.0/UDP 76.xx.xx.6:1025;branch=z9hG4bK7abf9ec38413c5df1.ff5fe84a8e81c1 27f
Max-Forwards: 70
From: “Ben” sip:103@216.xx.xx.82:5060;tag=a684eb1a81
To: “90845XXXXXX” sip:90845XXXXXX@216.xx.xx.82:5060;tag=as09eac3eb
Call-ID: ea2d1c8e24bddc7a
CSeq: 4627 ACK
User-Agent: Aastra 53i/2.5.2.1010
Content-Length: 0
<------------->
— (9 headers 0 lines) —
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
Retransmitting #3 (NAT) to 76.xx.xx.6:49904:
OPTIONS sip:104@76.xx.xx.6:49904;rinstance=aec6ef727a91733d SIP/2.0
Via: SIP/2.0/UDP 216.xx.xx.82:5060;branch=z9hG4bK266b980a;rport
From: “Unknown” sip:Unknown@216.xx.xx.82;tag=as09352ae6
To: sip:104@76.xx.xx.6:49904;rinstance=aec6ef727a91733d
Contact: sip:Unknown@216.xx.xx.82
Call-ID: 3090a11a4da324a00a8fec6b12e3403f@216.xx.xx.82
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 21 Sep 2010 07:34:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0