Outbound call is drop once a tone is received

I have a Trixbox with Version 2.8.0.4, FreePBX Version 2.7.0 and an Asterisk Version 1.6.0.26 running on Centos Release 5.5 (Final).

I’m not sure if this a FreePBX or Asterisk information issue but I would like to here your suggestions.

I’m currently using a Cisco SPA504G as my devices for my extensions.

Here is the scenario of my problem:

  1. I will dial an outgoing line from my extension and dial a 1800 number
  2. I will be able to connect to the 1800 number and I will hear the announcement asking me to provide information. The 1800 number is a Meet Me conference number and I will be able to enter the passcode to the conference but then when the message “After the tone please state your name” and when the tone is heard the line gets drop.

I’m using a Draytek 3300V as my FXO port and I have configured this without any problem for my 10 POTS. I can also use the FXO ports without any problem when I dial any number except when I receive any tone from the receiving party.

Appreciate your help. Thanks

Kindly post your config and logs. Are you recording voice simply into a local file or you have configured some sort of speech recognition system, as it is very populare and some voip service providers like Axvoice, are already planning to utilize speech technology in their voip system. Also verify that MeetMe is installed and running properly as this could happen. By inspecting dialplan logs the issue can be traced.

Hello CarolClark,

Thank you for answering my posts. Just to explain further, as explained in my previous post, I’m dialing a 1800 number thru my extension via an FXO port extension. I’m not using any voice recording or some sort of speech recognition as I’m not dialing into my own MeetMe Conference extension. My MeetMe is installed and running properly.

The 1800 number I’m dialing into also has MeetMe prompts which ask for a participant passcode (I’m not sure if this is also an Asterisk based system). I will be able to enter the pass code, but when I receive a tone the line gets dropped. If I use an analog telephone connected to my legacy Panasonic PBX, i don’t experience the same kind of problem.

As for the logs, I can only provide at the moment a capture of my verbose logs from the start of dialing the FXO extension until the call hangs up when I hear the tone, I’m not really familiar on how to properly capture the logs from the dial plan:

== Parsing ‘/etc/asterisk/asterisk.conf’: == Found
== Parsing ‘/etc/asterisk/extconfig.conf’: == Found
Connected to Asterisk 1.6.0.26-FONCORE-r78 currently running on dsctrixbox (pid = 3205)
Verbosity was 6 and is now 7
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [[email protected]:1] Macro(“SIP/500-00000bfa”, “exten-vm,novm,305”) in new stack
– Executing [[email protected]:1] Macro(“SIP/500-00000bfa”, “user-callerid,”) in new stack
– Executing [[email protected]:1] Set(“SIP/500-00000bfa”, “AMPUSER=500”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/500-00000bfa”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/500-00000bfa”, “1?Set(REALCALLERIDNUM=500)”) in new stack
– Executing [[email protected]:4] Set(“SIP/500-00000bfa”, “AMPUSER=500”) in new stack
– Executing [[email protected]:5] Set(“SIP/500-00000bfa”, “AMPUSERCIDNAME=Noel Flores SIP”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/500-00000bfa”, “0?report”) in new stack
– Executing [[email protected]:7] Set(“SIP/500-00000bfa”, “AMPUSERCID=500”) in new stack
– Executing [[email protected]:8] Set(“SIP/500-00000bfa”, “CALLERID(all)=“Noel Flores SIP” <500>”) in new stack
– Executing [[email protected]:9] ExecIf(“SIP/500-00000bfa”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [[email protected]:10] GotoIf(“SIP/500-00000bfa”, “0?continue”) in new stack
– Executing [[email protected]:11] Set(“SIP/500-00000bfa”, “__TTL=64”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/500-00000bfa”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [[email protected]:19] Set(“SIP/500-00000bfa”, “CALLERID(number)=500”) in new stack
– Executing [[email protected]:20] Set(“SIP/500-00000bfa”, “CALLERID(name)=Noel Flores SIP”) in new stack
– Executing [[email protected]rid:21] NoOp(“SIP/500-00000bfa”, “Using CallerID “Noel Flores SIP” <500>”) in new stack
– Executing [[email protected]:2] Set(“SIP/500-00000bfa”, “RingGroupMethod=none”) in new stack
– Executing [[email protected]:3] Set(“SIP/500-00000bfa”, “VMBOX=novm”) in new stack
– Executing [[email protected]:4] Set(“SIP/500-00000bfa”, “EXTTOCALL=305”) in new stack
– Executing [[email protected]:5] Set(“SIP/500-00000bfa”, “CFUEXT=”) in new stack
– Executing [[email protected]:6] Set(“SIP/500-00000bfa”, “CFBEXT=”) in new stack
– Executing [[email protected]:7] Set(“SIP/500-00000bfa”, “RT=”"") in new stack
– Executing [[email protected]:8] Macro(“SIP/500-00000bfa”, “record-enable,305,IN”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/500-00000bfa”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] ExecIf(“SIP/500-00000bfa”, “0?MacroExit()”) in new stack
– Executing [[email protected]:5] GotoIf(“SIP/500-00000bfa”, “0?Group:OUT”) in new stack
– Goto (macro-record-enable,s,15)
– Executing [[email protected]:15] GotoIf(“SIP/500-00000bfa”, “1?IN”) in new stack
– Goto (macro-record-enable,s,20)
– Executing [[email protected]:20] ExecIf(“SIP/500-00000bfa”, “0?MacroExit()”) in new stack
– Executing [[email protected]:21] NoOp(“SIP/500-00000bfa”, “Recording enable for 305”) in new stack
– Executing [[email protected]:22] Set(“SIP/500-00000bfa”, “CALLFILENAME=20120110-124004-1326170404.5859”) in new stack
– Executing [[email protected]:23] MixMonitor(“SIP/500-00000bfa”, “20120110-124004-1326170404.5859.wav,”) in new stack
– Executing [[email protected]:24] MacroExit(“SIP/500-00000bfa”, “”) in new stack
– Executing [[email protected]:9] Macro(“SIP/500-00000bfa”, “dial,”",tr,305") in new stack
== Begin MixMonitor Recording SIP/500-00000bfa
– Executing [[email protected]:1] GotoIf(“SIP/500-00000bfa”, “1?dial”) in new stack
– Goto (macro-dial,s,3)
– Executing [[email protected]:3] AGI(“SIP/500-00000bfa”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘Noel Flores SIP’ number is ‘500’

dialparties.agi: USE_CONFIRMATION: 'FALSE’
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 305 to extension map
– dialparties.agi: Extension 305 cf is disabled
– dialparties.agi: Extension 305 do not disturb is disabled
dialparties.agi: extnum 305 has: cw: 1; hascfb: 0 [] hascfu: 0 []
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
– dialparties.agi: dbset CALLTRACE/305 to 500
– dialparties.agi: Filtered ARG3: 305
– <SIP/500-00000bfa>AGI Script dialparties.agi completed, returning 0
– Executing [[email protected]:7] Dial(“SIP/500-00000bfa”, “SIP/305,”",tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called 305
– SIP/305-00000bfb is ringing
– SIP/305-00000bfb answered SIP/500-00000bfa
– Executing [[email protected]:1] Macro(“SIP/500-00000bfa”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/500-00000bfa”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/500-00000bfa”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [[email protected]:7] GotoIf(“SIP/500-00000bfa”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] Hangup(“SIP/500-00000bfa”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/500-00000bfa’ in macro ‘hangupcall’
== Spawn extension (macro-dial, h, 1) exited non-zero on ‘SIP/500-00000bfa’
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/500-00000bfa’ in macro ‘dial’
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on ‘SIP/500-00000bfa’ in macro ‘exten-vm’
== Spawn extension (from-internal, 305, 1) exited non-zero on ‘SIP/500-00000bfa’
– Executing [[email protected]:1] Macro(“SIP/500-00000bfa”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/500-00000bfa”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/500-00000bfa”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [[email protected]:7] GotoIf(“SIP/500-00000bfa”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] Hangup(“SIP/500-00000bfa”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/500-00000bfa’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/500-00000bfa’
== MixMonitor close filestream
== End MixMonitor Recording SIP/500-00000bfa

This is my asterisk.conf

directories ; remove the (!) to enable this
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk

;[options]
;verbose = 3
;debug = 3
;alwaysfork = yes ; same as -F at startup
;nofork = yes ; same as -f at startup
;quiet = yes ; same as -q at startup
;timestamp = yes ; same as -T at startup
;execincludes = yes ; support #exec in config files
;console = yes ; Run as console (same as -c at startup)
;highpriority = yes ; Run realtime priority (same as -p at startup)
;initcrypto = yes ; Initialize crypto keys (same as -i at startup)
;nocolor = yes ; Disable console colors
;dontwarn = yes ; Disable some warnings
;dumpcore = yes ; Dump core on crash (same as -g at startup)
;languageprefix = yes ; Use the new sound prefix path syntax
internal_timing = yes
;systemname = my_system_name ; prefix uniqueid with a system name for global uniqueness issues
;autosystemname = yes ; automatically set systemname to hostname - uses ‘localhost’ on failure, or systemname if set
;maxcalls = 10 ; Maximum amount of calls allowed
;maxload = 0.9 ; Asterisk stops accepting new calls if the load average exceed this limit
;maxfiles = 1000 ; Maximum amount of openfiles
;minmemfree = 1 ; in MBs, Asterisk stops accepting new calls if the amount of free memory falls below this watermark
;cache_record_files = yes ; Cache recorded sound files to another directory during recording
;record_cache_dir = /tmp ; Specify cache directory (used in conjunction with cache_record_files)
;transmit_silence_during_record = yes ; Transmit SLINEAR silence while a channel is being recorded
;transmit_silence = yes ; Transmit silence while a channel is in a waiting state, a recording only state, or when DTMF is
; being generated. Note that the silence internally is generated in raw signed linear format.
; This means that it must be transcoded into the native format of the channel before it can be sent
; to the device. It is for this reason that this is optional, as it may result in requiring a
; temporary codec translation path for a channel that may not otherwise require one.
;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of directly
;runuser = asterisk ; The user to run as
;rungroup = asterisk ; The group to run as

; Changing the following lines may compromise your security.
;[files]
;astctlpermissions = 0660
;astctlowner = root
;astctlgroup = apache
;astctl = asterisk.ctl

[compat]
pbx_realtime=1.6
res_agi=1.6
app_set=1.6

This is my chan_dahdi.conf:

;
; DAHDI telephony
;
; 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

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
;echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
busycount=6
callprogress=no
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include setup-pstn configs
#include dahdi-channels.conf

group=1

;Include PBXconfig configs
#include chan_dahdi_additional.conf

Anyone who can help me on my problem?