Voxilla VoIP Forum

Go Back   Voxilla VoIP Forum > Hardware/Software Support Forums > Asterisk Support Forum

Asterisk Support Forum Technical support, how-to guides, troubleshooting, and general assistance, from beginner to seasoned pro, this is where to discuss Asterisk, the most powerful open source PBX.


Closed Thread
 
LinkBack Thread Tools Rate Thread Display Modes
  #1 (permalink)  
Old December 22nd, 2005, 06:03 PM
Member
 
Join Date: Aug 2005
Posts: 62
degsod
Default Urgent Help Required

Hi I have just gone live with asterisk@home2.2 with digium Te110P Pri rate card UK ISDN provided by BT with 8 channels active.

I can make calls and receive calls ok then after about 14 outgoing calls the next call i make is always busy and then the next call I get the outoing ciruits are busy message.

I have enabled full logging (see below) the last number that succeeded was 02075551234, the next number dialed 02926661234 and all subsequant numbers dialed after that have the all circuits are busy message.

I have had bt monitoring the line at at the point of failure they say that there was no destination number coming down the line, asterisk logs seem to say that the outgoing number was dialed though?

I apoligise in advance for the length of the post, any help would be appreciated.

Regards


Derek

ZAPATA.CONF FILE:-

; Configuration file

[trunkgroups]

[channels]

language=en
context=from-pstn
;signalling=fxs_ks
signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=euroisdn

rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes
callerid=asreceived

usecallerid=yes
nationalprefix=0
internationalprefix=00
callerid=asreceived
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no


;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include genzaptelconf configs
#include zapata-auto.conf

;Include AMP configs
#include zapata_additional.conf
channel => 1-8

ZAPTEL.CONF FILE:-

# Global data

span=1,0,0,ccs,hdb3,crc4
bchan=1-8
dchan=16

loadzone=uk
defaultzone=uk


DEBUG FILE LAST 3 CALLS:-

Dec 22 17:32:25 VERBOSE[3721] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/5010-ad44'
Dec 22 17:32:25 DEBUG[3721] chan_sip.c: update_call_counter(5010) - decrement call limit counter
Dec 22 17:32:28 DEBUG[3161] acl.c: ##### Testing 10.0.0.141 with 10.0.0.0
Dec 22 17:32:28 DEBUG[3161] chan_sip.c: Setting NAT on RTP to 0
Dec 22 17:32:28 DEBUG[3161] chan_sip.c: Stopping retransmission on 'BA74CF67-CD2A-403C-8946-052B853B1E4D@10.0.0.141' of Response 1098: Match Found
Dec 22 17:32:28 DEBUG[3161] chan_sip.c: Setting NAT on RTP to 0
Dec 22 17:32:28 DEBUG[3161] chan_sip.c: Checking SIP call limits for device 5010
Dec 22 17:32:28 DEBUG[3161] chan_sip.c: build_route: Contact hop:
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Executing Macro("SIP/5010-c565", "dialout-trunk|1|2075551234|") in new stack
Dec 22 17:32:28 DEBUG[3730] pbx.c: Expression result is '1'
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Executing GotoIf("SIP/5010-c565", "1?3:2)") in new stack
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Goto (macro-dialout-trunk,s,3)
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Executing Macro("SIP/5010-c565", "user-callerid") in new stack
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Executing DBget("SIP/5010-c565", "AMPUSER=DEVICE/5010/user") in new stack
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- DBget: varname=AMPUSER, family=DEVICE, key=5010/user
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- DBget: set variable AMPUSER to 5010
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Executing DBget("SIP/5010-c565", "AMPUSERCIDNAME=AMPUSER/5010/cidname") in new stack
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=5010/cidname
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- DBget: set variable AMPUSERCIDNAME to A & B Support Desk
Dec 22 17:32:28 DEBUG[3730] pbx.c: Expression result is '0'
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Executing GotoIf("SIP/5010-c565", "0?5") in new stack
Dec 22 17:32:28 DEBUG[3730] pbx.c: Not taking any branch
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Executing SetCallerID("SIP/5010-c565", ""A &amp; B Support Desk" <5010>") in new stack
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Executing NoOp("SIP/5010-c565", "Using CallerID "A &amp; B Support Desk" <5010>") in new stack
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Executing Macro("SIP/5010-c565", "record-enable|5010|OUT") in new stack
Dec 22 17:32:28 DEBUG[3730] pbx.c: Function result is '0'
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Executing GotoIf("SIP/5010-c565", "0 > 0?2:4") in new stack
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Goto (macro-record-enable,s,4)
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Executing AGI("SIP/5010-c565", "recordingcheck|20051222-173228|1135272748.28") in new stack
Dec 22 17:32:28 VERBOSE[3730] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
Dec 22 17:32:29 VERBOSE[3730] logger.c: recordingcheck|20051222-173228|1135272748.28: Outbound recording not enabled
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- AGI Script recordingcheck completed, returning 0
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing NoOp("SIP/5010-c565", "No recording needed") in new stack
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing Macro("SIP/5010-c565", "outbound-callerid|1") in new stack
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing DBget("SIP/5010-c565", "USEROUTCID=AMPUSER/5010/outboundcid") in new stack
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- DBget: varname=USEROUTCID, family=AMPUSER, key=5010/outboundcid
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- DBget: set variable USEROUTCID to
Dec 22 17:32:29 DEBUG[3730] pbx.c: Expression result is '1'
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing GotoIf("SIP/5010-c565", "1?4") in new stack
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Goto (macro-outbound-callerid,s,4)
Dec 22 17:32:29 DEBUG[3730] pbx.c: Expression result is '1'
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing GotoIf("SIP/5010-c565", "1?6") in new stack
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Goto (macro-outbound-callerid,s,6)
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing NoOp("SIP/5010-c565", "CallerID set to "A &amp; B Support Desk" <5010>") in new stack
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing SetGroup("SIP/5010-c565", "OUT_1") in new stack
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing CheckGroup("SIP/5010-c565", "") in new stack
Dec 22 17:32:29 WARNING[3730] app_groupcount.c: CheckGroup requires an argument(max[@category][|options])
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing SetVar("SIP/5010-c565", "DIAL_NUMBER=2075551234") in new stack
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing SetVar("SIP/5010-c565", "DIAL_TRUNK=1") in new stack
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing AGI("SIP/5010-c565", "fixlocalprefix") in new stack
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
Dec 22 17:32:29 VERBOSE[3730] logger.c: fixlocalprefix: Could not open /etc/asterisk/localprefixes.conf
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- AGI Script fixlocalprefix completed, returning 0
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing SetVar("SIP/5010-c565", "OUTNUM=2075551234") in new stack
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing Cut("SIP/5010-c565", "custom=OUT_1|:|1") in new stack
Dec 22 17:32:29 WARNING[3730] ast_expr2.y: non-numeric argument
Dec 22 17:32:29 DEBUG[3730] pbx.c: Expression result is '0'
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing GotoIf("SIP/5010-c565", "0?16") in new stack
Dec 22 17:32:29 DEBUG[3730] pbx.c: Not taking any branch
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Executing Dial("SIP/5010-c565", "ZAP/g0/2075551234|120|TW") in new stack
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Requested transfer capability: 0x00 - SPEECH
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Called g0/2075551234
Dec 22 17:32:29 DEBUG[3163] chan_zap.c: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/1 span 1
Dec 22 17:32:29 VERBOSE[3730] logger.c: -- Zap/1-1 is proceeding passing it to SIP/5010-c565
Dec 22 17:32:31 VERBOSE[3163] logger.c: -- Channel 0/1, span 1 got hangup request
Dec 22 17:32:31 VERBOSE[3730] logger.c: -- Zap/1-1 is busy
Dec 22 17:32:31 DEBUG[3730] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Dec 22 17:32:31 DEBUG[3730] chan_zap.c: Hangup: channel: 1 index = 0, normal = 19, callwait = -1, thirdcall = -1
Dec 22 17:32:31 DEBUG[3730] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call
Dec 22 17:32:31 DEBUG[3730] chan_zap.c: disabled echo cancellation on channel 1
Dec 22 17:32:31 DEBUG[3730] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1
Dec 22 17:32:31 DEBUG[3730] chan_zap.c: Updated conferencing on 1, with 0 conference users
Dec 22 17:32:31 DEBUG[3730] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/1-1
Dec 22 17:32:31 DEBUG[3730] chan_zap.c: disabled echo cancellation on channel 1
Dec 22 17:32:31 VERBOSE[3730] logger.c: -- Hungup 'Zap/1-1'
Dec 22 17:32:31 VERBOSE[3730] logger.c: == Everyone is busy/congested at this time (1:1/0/0)
Dec 22 17:32:31 DEBUG[3730] app_dial.c: Exiting with DIALSTATUS=BUSY.
Dec 22 17:32:31 VERBOSE[3730] logger.c: -- Executing Goto("SIP/5010-c565", "s-BUSY|1") in new stack
Dec 22 17:32:31 VERBOSE[3730] logger.c: -- Goto (macro-dialout-trunk,s-BUSY,1)
Dec 22 17:32:31 VERBOSE[3730] logger.c: -- Executing NoOp("SIP/5010-c565", "Trunk is reporting BUSY") in new stack
Dec 22 17:32:31 VERBOSE[3730] logger.c: -- Executing Busy("SIP/5010-c565", "") in new stack
Dec 22 17:32:31 VERBOSE[3730] logger.c: == Spawn extension (macro-dialout-trunk, s-BUSY, 2) exited non-zero on 'SIP/5010-c565' in macro 'dialout-trunk'
Dec 22 17:32:31 VERBOSE[3730] logger.c: == Spawn extension (from-internal, 902075551234, 1) exited non-zero on 'SIP/5010-c565'
Dec 22 17:32:31 VERBOSE[3730] logger.c: -- Executing Macro("SIP/5010-c565", "hangupcall") in new stack
Dec 22 17:32:31 VERBOSE[3730] logger.c: -- Executing ResetCDR("SIP/5010-c565", "w") in new stack
Dec 22 17:32:31 DEBUG[3161] chan_sip.c: Stopping retransmission on 'BA74CF67-CD2A-403C-8946-052B853B1E4D@10.0.0.141' of Response 1099: Match Not Found
Dec 22 17:32:31 DEBUG[3730] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Dec 22 17:32:31 DEBUG[3730] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel ,lastapp,lastdata,duration,billsec,disposition,ama flags,accountcode,uniqueid) VALUES ('2005-12-22 17:32:28','\"A &amp; B Support Desk\" <5010>','5010','902075551234','from-internal', 'SIP/5010-c565','Zap/1-1','ResetCDR','w',3,0,'BUSY',3,'','1135272748.28')
Dec 22 17:32:31 VERBOSE[3163] logger.c: == Setting global variable 'USERUSERINFO' to '?( '
Dec 22 17:32:31 VERBOSE[3730] logger.c: -- Executing NoCDR("SIP/5010-c565", "") in new stack
Dec 22 17:32:31 WARNING[3730] cdr.c: CDR on channel 'SIP/5010-c565' not posted
Dec 22 17:32:31 WARNING[3730] cdr.c: CDR on channel 'SIP/5010-c565' lacks end
Dec 22 17:32:31 VERBOSE[3730] logger.c: -- Executing Wait("SIP/5010-c565", "5") in new stack
Dec 22 17:32:31 VERBOSE[3730] logger.c: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/5010-c565' in macro 'hangupcall'
Dec 22 17:32:31 VERBOSE[3730] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/5010-c565'
Dec 22 17:32:31 DEBUG[3730] chan_sip.c: update_call_counter(5010) - decrement call limit counter
Dec 22 17:32:40 DEBUG[3161] acl.c: ##### Testing 10.0.0.141 with 10.0.0.0
Dec 22 17:32:40 DEBUG[3161] chan_sip.c: Setting NAT on RTP to 0
Dec 22 17:32:40 DEBUG[3161] chan_sip.c: Stopping retransmission on 'F46706FD-C229-4EED-A1E6-D74694C7B8F1@10.0.0.141' of Response 6716: Match Found
Dec 22 17:32:40 DEBUG[3161] chan_sip.c: Setting NAT on RTP to 0
Dec 22 17:32:40 DEBUG[3161] chan_sip.c: Checking SIP call limits for device 5010
Dec 22 17:32:40 DEBUG[3161] chan_sip.c: build_route: Contact hop:
Dec 22 17:32:40 VERBOSE[3739] logger.c: -- Executing Macro("SIP/5010-f823", "dialout-trunk|1|2926661234|") in new stack
Dec 22 17:32:40 DEBUG[3739] pbx.c: Expression result is '1'
Dec 22 17:32:40 VERBOSE[3739] logger.c: -- Executing GotoIf("SIP/5010-f823", "1?3:2)") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Goto (macro-dialout-trunk,s,3)
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing Macro("SIP/5010-f823", "user-callerid") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing DBget("SIP/5010-f823", "AMPUSER=DEVICE/5010/user") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- DBget: varname=AMPUSER, family=DEVICE, key=5010/user
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- DBget: set variable AMPUSER to 5010
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing DBget("SIP/5010-f823", "AMPUSERCIDNAME=AMPUSER/5010/cidname") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=5010/cidname
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- DBget: set variable AMPUSERCIDNAME to A &amp; B Support Desk
Dec 22 17:32:41 DEBUG[3739] pbx.c: Expression result is '0'
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing GotoIf("SIP/5010-f823", "0?5") in new stack
Dec 22 17:32:41 DEBUG[3739] pbx.c: Not taking any branch
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing SetCallerID("SIP/5010-f823", ""A &amp; B Support Desk" <5010>") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing NoOp("SIP/5010-f823", "Using CallerID "A &amp; B Support Desk" <5010>") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing Macro("SIP/5010-f823", "record-enable|5010|OUT") in new stack
Dec 22 17:32:41 DEBUG[3739] pbx.c: Function result is '0'
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing GotoIf("SIP/5010-f823", "0 > 0?2:4") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Goto (macro-record-enable,s,4)
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing AGI("SIP/5010-f823", "recordingcheck|20051222-173241|1135272760.30") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
Dec 22 17:32:41 VERBOSE[3739] logger.c: recordingcheck|20051222-173241|1135272760.30: Outbound recording not enabled
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- AGI Script recordingcheck completed, returning 0
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing NoOp("SIP/5010-f823", "No recording needed") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing Macro("SIP/5010-f823", "outbound-callerid|1") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing DBget("SIP/5010-f823", "USEROUTCID=AMPUSER/5010/outboundcid") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- DBget: varname=USEROUTCID, family=AMPUSER, key=5010/outboundcid
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- DBget: set variable USEROUTCID to
Dec 22 17:32:41 DEBUG[3739] pbx.c: Expression result is '1'
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing GotoIf("SIP/5010-f823", "1?4") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Goto (macro-outbound-callerid,s,4)
Dec 22 17:32:41 DEBUG[3739] pbx.c: Expression result is '1'
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing GotoIf("SIP/5010-f823", "1?6") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Goto (macro-outbound-callerid,s,6)
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing NoOp("SIP/5010-f823", "CallerID set to "A &amp; B Support Desk" <5010>") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing SetGroup("SIP/5010-f823", "OUT_1") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing CheckGroup("SIP/5010-f823", "") in new stack
Dec 22 17:32:41 WARNING[3739] app_groupcount.c: CheckGroup requires an argument(max[@category][|options])
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing SetVar("SIP/5010-f823", "DIAL_NUMBER=2926661234") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing SetVar("SIP/5010-f823", "DIAL_TRUNK=1") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing AGI("SIP/5010-f823", "fixlocalprefix") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
Dec 22 17:32:41 VERBOSE[3739] logger.c: fixlocalprefix: Could not open /etc/asterisk/localprefixes.conf
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- AGI Script fixlocalprefix completed, returning 0
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing SetVar("SIP/5010-f823", "OUTNUM=2926661234") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing Cut("SIP/5010-f823", "custom=OUT_1|:|1") in new stack
Dec 22 17:32:41 WARNING[3739] ast_expr2.y: non-numeric argument
Dec 22 17:32:41 DEBUG[3739] pbx.c: Expression result is '0'
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing GotoIf("SIP/5010-f823", "0?16") in new stack
Dec 22 17:32:41 DEBUG[3739] pbx.c: Not taking any branch
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing Dial("SIP/5010-f823", "ZAP/g0/2926661234|120|TW") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Requested transfer capability: 0x00 - SPEECH
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Called g0/2926661234
Dec 22 17:32:41 VERBOSE[3163] logger.c: -- Channel 0/1, span 1 got hangup
Dec 22 17:32:41 DEBUG[3739] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Dec 22 17:32:41 DEBUG[3739] chan_zap.c: Hangup: channel: 1 index = 0, normal = 19, callwait = -1, thirdcall = -1
Dec 22 17:32:41 DEBUG[3739] chan_zap.c: Already hungup... Calling hangup once, and clearing call
Dec 22 17:32:41 DEBUG[3739] chan_zap.c: disabled echo cancellation on channel 1
Dec 22 17:32:41 DEBUG[3739] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1
Dec 22 17:32:41 DEBUG[3739] chan_zap.c: Updated conferencing on 1, with 0 conference users
Dec 22 17:32:41 DEBUG[3739] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/1-1
Dec 22 17:32:41 DEBUG[3739] chan_zap.c: disabled echo cancellation on channel 1
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Hungup 'Zap/1-1'
Dec 22 17:32:41 VERBOSE[3739] logger.c: == No one is available to answer at this time (1:0/0/0)
Dec 22 17:32:41 DEBUG[3739] app_dial.c: Exiting with DIALSTATUS=NOANSWER.
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing Goto("SIP/5010-f823", "s-NOANSWER|1") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Goto (macro-dialout-trunk,s-NOANSWER,1)
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing NoOp("SIP/5010-f823", "Dial failed due to NOANSWER") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing Macro("SIP/5010-f823", "outisbusy") in new stack
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Executing Playback("SIP/5010-f823", "all-circuits-busy-now") in new stack
Dec 22 17:32:41 DEBUG[3739] channel.c: Scheduling timer at 160 sample intervals
Dec 22 17:32:41 VERBOSE[3739] logger.c: -- Playing 'all-circuits-busy-now' (language 'en')
Dec 22 17:32:41 DEBUG[3161] chan_sip.c: Stopping retransmission on 'F46706FD-C229-4EED-A1E6-D74694C7B8F1@10.0.0.141' of Response 6717: Match Found
Dec 22 17:32:42 DEBUG[3739] channel.c: Scheduling timer at 0 sample intervals
Dec 22 17:32:42 VERBOSE[3739] logger.c: == Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/5010-f823' in macro 'outisbusy'
Dec 22 17:32:42 VERBOSE[3739] logger.c: == Spawn extension (from-internal, 902926661234, 2) exited non-zero on 'SIP/5010-f823'
Dec 22 17:32:42 VERBOSE[3739] logger.c: -- Executing Macro("SIP/5010-f823", "hangupcall") in new stack
Dec 22 17:32:42 VERBOSE[3739] logger.c: -- Executing ResetCDR("SIP/5010-f823", "w") in new stack
Dec 22 17:32:42 DEBUG[3739] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Dec 22 17:32:42 DEBUG[3739] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel ,lastapp,lastdata,duration,billsec,disposition,ama flags,accountcode,uniqueid) VALUES ('2005-12-22 17:32:40','\"A &amp; B Support Desk\" <5010>','5010','902926661234','from-internal', 'SIP/5010-f823','Zap/1-1','ResetCDR','w',2,1,'ANSWERED',3,'','1135272760. 30')
Dec 22 17:32:42 VERBOSE[3739] logger.c: -- Executing NoCDR("SIP/5010-f823", "") in new stack
Dec 22 17:32:42 WARNING[3739] cdr.c: CDR on channel 'SIP/5010-f823' not posted
Dec 22 17:32:42 WARNING[3739] cdr.c: CDR on channel 'SIP/5010-f823' lacks end
Dec 22 17:32:42 VERBOSE[3739] logger.c: -- Executing Wait("SIP/5010-f823", "5") in new stack
Dec 22 17:32:42 VERBOSE[3739] logger.c: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/5010-f823' in macro 'hangupcall'
Dec 22 17:32:42 VERBOSE[3739] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/5010-f823'
Dec 22 17:32:42 DEBUG[3739] chan_sip.c: update_call_counter(5010) - decrement call limit counter
Dec 22 17:32:53 DEBUG[3309] manager.c: Manager received command 'Command'
Dec 22 17:32:53 DEBUG[3309] manager.c: Manager received command 'Command'
Dec 22 17:34:48 DEBUG[3161] acl.c: ##### Testing 10.0.0.203 with 10.0.0.0
Dec 22 17:34:49 DEBUG[3161] chan_sip.c: Stopping retransmission on '3c26700c57e4-s6ctc580vq0c@snom360' of Request 102: Match Found
Dec 22 17:34:53 DEBUG[3309] manager.c: Manager received command 'Command'
Dec 22 17:34:53 DEBUG[3309] manager.c: Manager received command 'Command'
Dec 22 17:35:04 DEBUG[3161] acl.c: ##### Testing 10.0.0.203 with 10.0.0.0
Dec 22 17:35:04 VERBOSE[3161] logger.c: -- Saved useragent "snom360/4.1" for peer 4001
Dec 22 17:35:04 DEBUG[3161] acl.c: ##### Testing 10.0.0.203 with 10.0.0.0
Dec 22 17:35:04 DEBUG[3161] chan_sip.c: Stopping retransmission on '3c26700c5a55-o6sg676q3eq3@snom360' of Request 102: Match Found
Dec 22 17:35:04 DEBUG[3161] acl.c: ##### Testing 10.0.0.203 with 10.0.0.0
Dec 22 17:35:05 DEBUG[3161] chan_sip.c: Stopping retransmission on '3c26700c5f37-5hh0i0j1shf1@snom360' of Request 102: Match Found
Dec 22 17:35:05 DEBUG[3161] acl.c: ##### Testing 10.0.0.203 with 10.0.0.0
Dec 22 17:35:05 DEBUG[3161] acl.c: ##### Testing 10.0.0.203 with 10.0.0.0
Dec 22 17:35:05 DEBUG[3161] chan_sip.c: Stopping retransmission on '3c26700c6419-ofnxgr2totko@snom360' of Request 102: Match Found
Dec 22 17:35:05 DEBUG[3161] chan_sip.c: Stopping retransmission on '3c26700c668a-554sycli06wo@snom360' of Request 102: Match Found
Dec 22 17:35:05 DEBUG[3161] acl.c: ##### Testing 10.0.0.203 with 10.0.0.0
Dec 22 17:35:05 DEBUG[3161] acl.c: ##### Testing 10.0.0.203 with 10.0.0.0
Dec 22 17:35:05 DEBUG[3161] chan_sip.c: Stopping retransmission on '3c26700c68fb-02gl5hc156tu@snom360' of Request 102: Match Found
Dec 22 17:35:05 DEBUG[3161] acl.c: ##### Testing 10.0.0.203 with 10.0.0.0
Dec 22 17:35:05 DEBUG[3161] chan_sip.c: Stopping retransmission on '3c26700c6ddd-gg4chtivnujo@snom360' of Request 102: Match Found
Dec 22 17:35:05 DEBUG[3161] chan_sip.c: Stopping retransmission on '3c26700c704e-pmcshkm0slhj@snom360' of Request 102: Match Found
Dec 22 17:35:06 DEBUG[3161] acl.c: ##### Testing 10.0.0.203 with 10.0.0.0
Dec 22 17:35:06 DEBUG[3161] chan_sip.c: Stopping retransmission on '3c26700c61a8-4g672j9hk5tw@snom360' of Request 102: Match Found
Dec 22 17:35:12 DEBUG[3161] acl.c: ##### Testing 10.0.0.203 with 10.0.0.0
Dec 22 17:35:12 DEBUG[3161] chan_sip.c: Stopping retransmission on '11dc903861783d7c7cd432d903aa72ce@10.0.0.221' of Request 102: Match Found
Dec 22 17:35:19 DEBUG[3161] chan_sip.c: Auto destroying call '3c267009a393-jxdtxrv2p2k8@snom360'
Dec 22 17:35:19 DEBUG[3161] chan_sip.c: Auto destroying call '3c267009a393-jxdtxrv2p2k8@snom360'
Digg this Post!Add Post to del.icio.usBookmark Post in TechnoratiFurl this Post!
Closed Thread

Thread Tools
Display Modes Rate This Thread
Rate This Thread:

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off
Trackbacks are On
Pingbacks are On
Refbacks are On

Similar Threads
Thread Thread Starter Forum Replies Last Post
Urgent request: differences SPA-3000/SPA-3102 carola Linksys (Sipura) VoIP Support Forum 42 October 10th, 2006 02:38 AM
URGENT - NEED DIAL PLAN FOR MEXICO with Linksys 2100 & 2 elmagito Linksys (Sipura) VoIP Support Forum 2 August 29th, 2006 09:44 PM
Support Required irfanfarooq Linksys (Sipura) VoIP Support Forum 1 February 9th, 2006 08:55 PM
URGENT! Removing Cent OS from my Asterisk box! simanu Asterisk Support Forum 4 October 6th, 2005 02:55 AM
Is TelAppliant DOWN?..Need help--Urgent rizsher Linksys (Sipura) VoIP Support Forum 1 August 5th, 2004 04:30 AM


Voxilla News

More Voxilla news



All times are GMT. The time now is 01:11 AM.


vBulletin, Copyright ©2000 - 2009, Jelsoft Enterprises Ltd.
SEO by vBSEO 3.2.0 ©2008, Crawlability, Inc.
Logos and trademarks are the property of Voxilla or their respective owner. All other content © 2003-2009 by Voxilla, Inc.