Dear Trixbox-ers,
I am having some strange problem, I am receiving empty calls, every 6-7 minutes, the phone rings for about 16-20 seconds, and hangs up.
This is pretty irritating.
Just that you know, I have ATA adaptors between my analog phone and Asterisk Trixbox box. Could it be the ATA adaptor?
Any help is appreciated.
Thanks in advance,
Ilir
Here is the debug info that I could extract while this weird behavior takes place:
I would appreciate any help! (10.20.30.200 is the IP of the asterisk box, and 10.20.30.154 is the ip of the ata adapter. )
— (8 headers 0 lines)—
Destroying call '2445a592330961cf14674dfb6b4ac3d1@10.20.30.200’
asterisk1*CLI>
<-- SIP read from 10.20.30.154:5060:
SIP/2.0 200 OK
From: Unknownsip:Unknown@10.20.30.200;tag=as2dcbbb6d
To: sip:101@10.20.30.154
Call-ID: 129f56797abb95f103c3914b764c7229@10.20.30.200
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 10.20.30.200:5060;branch=z9hG4bK35c154ba
User-Agent: Myson Century/VR4.1 Build-Date Dec 15 2005
Content-Length: 0
— (8 headers 0 lines)—
Destroying call '129f56797abb95f103c3914b764c7229@10.20.30.200’
Destroying call 'DBTk-dkL0Pv049@10.20.30.200’
Destroying call '3gpvu0-trZ0Pv049@10.20.30.200’
Destroying call ‘0o0830-ZSM0Pv049@10.20.30.200’
– Starting simple switch on ‘Zap/4-1’
– Executing NoOp(“Zap/4-1”, "Entering from-zaptel with DID == ") in new stack
– Executing Set(“Zap/4-1”, “DID=s”) in new stack
– Executing NoOp(“Zap/4-1”, “DID is now s”) in new stack
– Executing GotoIf(“Zap/4-1”, “1?zapok:notzap”) in new stack
– Goto (from-zaptel,s,7)
– Executing NoOp(“Zap/4-1”, “Is a Zaptel Channel”) in new stack
– Executing Set(“Zap/4-1”, “CHAN=4-1”) in new stack
– Executing Set(“Zap/4-1”, “CHAN=4”) in new stack
– Executing Macro(“Zap/4-1”, “from-zaptel-4|s|1”) in new stack
– Executing NoOp(“Zap/4-1”, “Entering macro-from-zaptel-4 with DID = s”) in new stack
– Executing Set(“Zap/4-1”, “FROM_DID=s”) in new stack
– Executing Goto(“Zap/4-1”, “ext-local|101|1”) in new stack
– Goto (ext-local,101,1)
== Channel ‘Zap/4-1’ jumping out of macro ‘from-zaptel-4’
– Executing Macro(“Zap/4-1”, “exten-vm|101|101”) in new stack
– Executing Macro(“Zap/4-1”, “user-callerid”) in new stack
– Executing GotoIf(“Zap/4-1”, “0?report”) in new stack
– Executing GotoIf(“Zap/4-1”, “0?start”) in new stack
– Executing Set(“Zap/4-1”, “REALCALLERIDNUM=”) in new stack
– Executing NoOp(“Zap/4-1”, "REALCALLERIDNUM is ") in new stack
– Executing Set(“Zap/4-1”, “AMPUSER=”) in new stack
– Executing Set(“Zap/4-1”, “AMPUSERCIDNAME=”) in new stack
– Executing GotoIf(“Zap/4-1”, “1?report”) in new stack
– Goto (macro-user-callerid,s,9)
– Executing NoOp(“Zap/4-1”, “Using CallerID “” <>”) in new stack
– Executing Set(“Zap/4-1”, “FROMCONTEXT=exten-vm”) in new stack
– Executing Set(“Zap/4-1”, “VMBOX=101”) in new stack
– Executing Set(“Zap/4-1”, “EXTTOCALL=101”) in new stack
– Executing Set(“Zap/4-1”, “CFUEXT=”) in new stack
– Executing Set(“Zap/4-1”, “RT=20”) in new stack
– Executing Macro(“Zap/4-1”, “record-enable|101|IN”) in new stack
– Executing GotoIf(“Zap/4-1”, “0 > 0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing AGI(“Zap/4-1”, “recordingcheck|20061226-134357|1167137029.23”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20061226-134357|1167137029.23: Inbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing NoOp(“Zap/4-1”, “No recording needed”) in new stack
– Executing GotoIf(“Zap/4-1”, “0?dolocaldial|1”) in new stack
– Executing Macro(“Zap/4-1”, “dial|20|tr|101”) in new stack
– Executing AGI(“Zap/4-1”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
– dialparties.agi: priority is 1
dialparties.agi: Caller ID name is ‘unknown’ number is 'unknown’
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 101 to extension map
– dialparties.agi: Extension 101 cf is disabled
– dialparties.agi: Extension 101 do not disturb is disabled
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
– dialparties.agi: Checking CW and CFB status for extension 101
– dialparties.agi: DbSet CALLTRACE/101 to unknown
– AGI Script dialparties.agi completed, returning 0
– Executing Dial(“Zap/4-1”, “SIP/101|20|tr”) in new stack
We’re at 10.20.30.200 port 13366
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 10.20.30.154:5060:
INVITE sip:101@10.20.30.154:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.200:5060;branch=z9hG4bK5f8623c6
From: “Unknown” sip:Unknown@10.20.30.200;tag=as235aaa37
To: sip:101@10.20.30.154:5060
Contact: sip:Unknown@10.20.30.200
Call-ID: 41c6ecc06f8f8ec81df919bc2a27543c@10.20.30.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 26 Dec 2006 12:43:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 7743 7743 IN IP4 10.20.30.200
s=session
c=IN IP4 10.20.30.200
t=0 0
m=audio 13366 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 - - - -
– Called 101
asterisk1*CLI>
<-- SIP read from 10.20.30.154:5060:
SIP/2.0 100 Trying
From: Unknownsip:Unknown@10.20.30.200;tag=as235aaa37
To: sip:101@10.20.30.154
Call-ID: 41c6ecc06f8f8ec81df919bc2a27543c@10.20.30.200
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.20.30.200:5060;branch=z9hG4bK5f8623c6
Content-Length: 0
— (7 headers 0 lines)—
asterisk1*CLI>
<-- SIP read from 10.20.30.154:5060:
SIP/2.0 180 Ringing
From: Unknownsip:Unknown@10.20.30.200;tag=as235aaa37
To: sip:101@10.20.30.154;tag=L4H59-I5OD0
Call-ID: 41c6ecc06f8f8ec81df919bc2a27543c@10.20.30.200
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.20.30.200:5060;branch=z9hG4bK5f8623c6
Contact: 101sip:101@10.20.30.154:5060
User-Agent: Myson Century/VR4.1 Build-Date Dec 15 2005
Content-Length: 0
— (9 headers 0 lines)—
– SIP/101-0934e498 is ringing
Scheduling destruction of call ‘41c6ecc06f8f8ec81df919bc2a27543c@10.20.30.200’ in 32000 ms
Reliably Transmitting (no NAT) to 10.20.30.154:5060:
CANCEL sip:101@10.20.30.154:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.200:5060;branch=z9hG4bK5f8623c6
From: “Unknown” sip:Unknown@10.20.30.200;tag=as235aaa37
To: sip:101@10.20.30.154:5060
Contact: sip:Unknown@10.20.30.200
Call-ID: 41c6ecc06f8f8ec81df919bc2a27543c@10.20.30.200
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
== Spawn extension (macro-dial, s, 10) exited non-zero on ‘Zap/4-1’ in macro ‘dial’
== Spawn extension (macro-dial, s, 10) exited non-zero on ‘Zap/4-1’ in macro ‘exten-vm’
== Spawn extension (macro-dial, s, 10) exited non-zero on ‘Zap/4-1’
– Hungup 'Zap/4-1’
asterisk1*CLI>
<-- SIP read from 10.20.30.154:5060:
SIP/2.0 200 OK
From: Unknownsip:Unknown@10.20.30.200;tag=as235aaa37
To: sip:101@10.20.30.154
Call-ID: 41c6ecc06f8f8ec81df919bc2a27543c@10.20.30.200
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 10.20.30.200:5060;branch=z9hG4bK5f8623c6
User-Agent: Myson Century/VR4.1 Build-Date Dec 15 2005
Content-Length: 0
— (8 headers 0 lines)—
asterisk1*CLI>
<-- SIP read from 10.20.30.154:5060:
SIP/2.0 487 Request Terminated
From: Unknownsip:Unknown@10.20.30.200;tag=as235aaa37
To: sip:101@10.20.30.154;tag=L4H59-I5OD0
Call-ID: 41c6ecc06f8f8ec81df919bc2a27543c@10.20.30.200
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.20.30.200:5060;branch=z9hG4bK5f8623c6
User-Agent: Myson Century/VR4.1 Build-Date Dec 15 2005
Content-Length: 0
— (8 headers 0 lines)—
Transmitting (no NAT) to 10.20.30.154:5060:
ACK sip:101@10.20.30.154:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.200:5060;branch=z9hG4bK5f8623c6
From: “Unknown” sip:Unknown@10.20.30.200;tag=as235aaa37
To: sip:101@10.20.30.154:5060;tag=L4H59-I5OD0
Contact: sip:Unknown@10.20.30.200
Call-ID: 41c6ecc06f8f8ec81df919bc2a27543c@10.20.30.200
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Destroying call ‘41c6ecc06f8f8ec81df919bc2a27543c@10.20.30.200’
I am really desperate for help. Now the empty calls keep the trunk busy for 30-31 seconds, and such “calls” are being generated every 6 minutes.
This is how my zapata.conf file looks like:
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes
busydetect=yes
busycount=4
usecallerid=yes
hidecallerid=no
;callwaiting=yes
;usecallingpres=yes
callwaitingcallerid=yes
;threewaycalling=yes
transfer=yes
;cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=3.0
txgain=4.0
group=0
callgroup=1
pickupgroup=1
immediate=no
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;Include genzaptelconf configs
#include zapata-auto.conf
i promise not to be offended 
the call would appear to be from the Zap channel you have connected, nothing to do with an ATA.
do you have any services on the line that might be affecting the voltage ? Zaptel is seeing a change to line status and indicating a ring … your incoming context is then taking the call and doing whatever you have in there with it.
can you put a meter across the line and see what happens … 6-7 minutes won’t be too long to watch a meter 
if nothing, you might have a faulty module or card. what card are you using ?
type of card TDM02B
Call Logs -
Calldate Channel Source Clid Dst Disposition Duration
2007-01-17 16:29:52 SIP/101-08... 101 "Arka I" <101> 110 ANSWERED 01:19
2007-01-17 16:27:59 SIP/101-08... 101 "Arka I" <101> 110 ANSWERED 00:58
2007-01-17 16:27:40 Zap/4-1... hang ANSWERED 01:00
2007-01-17 16:21:17 Zap/4-1... 038246888 038246888 101 ANSWERED 01:49
2007-01-17 16:20:29 SIP/101-08... 101 "Arka I" <101> 116 ANSWERED 00:35
2007-01-17 16:19:44 SIP/101-08... 101 "Arka I" <101> 110 ANSWERED 00:31
2007-01-17 16:16:04 Zap/4-1... hang ANSWERED 01:00
2007-01-17 16:14:30 SIP/101-08... 101 "Arka I" <101> 116 ANSWERED 04:43
2007-01-17 16:10:12 Zap/4-1... hang ANSWERED 01:00
2007-01-17 16:04:24 Zap/4-1... hang ANSWERED 01:00
2007-01-17 16:03:39 SIP/110-08... 110 "Depo I" <110> 103 ANSWERED 02:22
2007-01-17 16:02:06 SIP/116-08... 116 "ilir-cacttus" <116> 103 ANSWERED 01:36
2007-01-17 15:58:32 Zap/4-1... hang ANSWERED 01:00
2007-01-17 15:54:22 SIP/116-08... 116 "ilir-cacttus" <116> 103 ANSWERED 00:42
2007-01-17 15:52:33 Zap/4-1... hang ANSWERED 00:59
2007-01-17 15:50:52 Zap/4-1... 038246888 038246888 101 ANSWERED 00:27
as you can see from report, calls number 3,7,9,10,11,13,15 are all empty calls that happen every 6 min or so, and occupy the Zap channel with the PSTN connection attached.
do you need to see any other conf. files, like zapata.conf or zaptel.conf or any output from asterisk CLI… ???
anyone has an idea about this? im realy desperate now :S
i thought you had an answer. if you’re really desperate surely you’d be checking using the suggestions given ?
well i didnt realy understand what you mean by “can you put a meter across the line and see what happens … 6-7 minutes won’t be too long to watch a meter” - you mean checking the current and voltage ?
I told you the type of the card. Is there a way to check if IT is the problem without replacing it (they do cost a lot :S).
the card is seeing a line condition (or thinks it is) that indicates ringing. so Asterisk is just doing what is expected of it.
you can either change the card, or get your multimeter out and see the line condition for yourself. if you see no increase in voltage or a polarity reversal then it must be the card. if you do see them, then there is something on the line that is causing it … time to ring the telco.