Can not make any outbound call (GoAutoDile)

Dear all,

Thank you for taking the time to read my message. I am currently using the well-known software called GoAutoDial. I had asked a question on their forum, but it seems the forum is not active anymore.

This software uses an asterisk in the background. I have set up my VoIP provider, and as you can see here:

Now when I log in to the agent, I can hear the message
" You are currently the only person in this conference" indicates that I have no problem on the server…“I think!”

When I make a call, it times out without connecting.

My Voip provider say :
we support regular SIP protocol, wich works on the Asterisk platform as well

Please advise!

cat /etc/asterisk/asterisk.conf
astetcdir => /etc/asterisk
astmoddir => /usr/lib64/asterisk/modules
astvarlibdir => /usr/share/asterisk
astdbdir => /var/spool/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /usr/share/asterisk
astagidir => /usr/share/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /run/asterisk
astlogdir => /var/log/asterisk
astsbindir => /usr/sbin

;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.
;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.
;mindtmfduration = 80           ; Set minimum DTMF duration in ms (default 80 ms)
                                ; If we get shorter DTMF messages, these will be
                                ; changed to the minimum duration
;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 = 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.
;lightbackground = yes          ; If your terminal is set for a light-colored
                                ; background.
;forceblackbackground = yes     ; Force the background of the terminal to be
                                ; black, in order for terminal colors to show
                                ; up properly.
;defaultlanguage = en           ; Default language
documentation_language = en_US  ; Set the language you want documentation
                                ; displayed in. Value is in the same format as
                                ; locale names.
;hideconnect = yes              ; Hide messages displayed when a remote console
                                ; connects and disconnects.
;lockconfdir = no               ; Protect the directory containing the
                                ; configuration files (/etc/asterisk) with a
                                ; lock.
;stdexten = gosub               ; How to invoke the extensions.conf stdexten.
                                ; macro - Invoke the stdexten using a macro as
                                ;         done by legacy Asterisk versions.
                                ; gosub - Invoke the stdexten using a gosub as
                                ;         documented in extensions.conf.sample.
                                ; Default gosub.
;live_dangerously = no          ; Enable the execution of 'dangerous' dialplan
                                ; functions from external sources (AMI,
                                ; etc.) These functions (such as SHELL) are
                                ; considered dangerous because they can allow
                                ; privilege escalation.
                                ; Default no
;entityid=00:11:22:33:44:55     ; Entity ID.
                                ; This is in the form of a MAC address.
                                ; It should be universally unique.
                                ; It must be unique between servers communicating
                                ; with a protocol that uses this value.
                                ; This is currently is used by DUNDi and
                                ; Exchanging Device and Mailbox State
                                ; using protocols: XMPP, Corosync and PJSIP.
;rtp_pt_dynamic = 96            ; Normally the Dynamic RTP Payload Type numbers
                                ; are 96-127, which allow 32 formats. When you
                                ; use more and receive the message "No Dynamic
                                ; RTP mapping available", extend the dynamic
                                ; range by going for 35 (or 0) instead of 96.
                                ; This allows 29 (or 64) more formats. 96 is the
                                ; default because any number below might be
                                ; rejected by a remote implementation; although
                                ; no such broken implementation is known, yet.

; Changing the following lines may compromise your security.
;astctlpermissions = 0660
;astctlowner = root
;astctlgroup = apache
;astctl = asterisk.ctl
cat /etc/asterisk/sip-goautodial.conf
;encryption=yes ;uncomment for TLS encryption
nat=force_rport,comedia ;change me to my FQDN

 cat /etc/asterisk/sip.conf
context=trunkinbound            ; Default context for incoming calls
allowguest=no                  ; Allow or reject guest calls (default is yes)
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld             ; Realm for digest authentication
bindport=5070                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=                ; IP address to bind to ( binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld            ; Set default domain for this host
;pedantic=yes                   ; Enable checking of tags in headers,
;tos_sip=cs3                    ; Sets TOS for SIP packets.
;tos_audio=ef                   ; Sets TOS for RTP audio packets.
;tos_video=af41                 ; Sets TOS for RTP video packets.
;maxexpiry=3600                 ; Maximum allowed time of incoming registrations
;minexpiry=60                   ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120              ; Default length of incoming/outgoing registration
;t1min=100                      ; Minimum roundtrip time for messages to monitored hosts
;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10                    ; Default time between mailbox checks for peers
;buggymwi=no                    ; Cisco SIP firmware doesn't support the MWI RFC
;vmexten=voicemail              ; dialplan extension to reach mailbox sets the
disallow=all                    ; First disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
language=en                     ; Default language setting for all users/peers
relaxdtmf=yes                   ; Relax dtmf handling
trustrpid = no                  ; If Remote-Party-ID should be trusted
sendrpid = yes                  ; If Remote-Party-ID should be sent
progressinband=no               ; If we should generate in-band ringing always
;useragent=Asterisk PBX         ; Allows you to change the user agent string
;promiscredir = no              ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no               ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833              ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes           ; send compact sip headers.
videosupport=no                 ; Turn on support for SIP video. You need to turn this on
;maxcallbitrate=384             ; Maximum bitrate for video calls (default 384 kb/s)
callevents=yes                  ; generate manager events when sip ua
;alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is to be rejected,
;g726nonstandard = yes          ; If the peer negotiates G726-32 audio, use AAL2 packing
;matchexterniplocally = yes     ; Only substitute the externip or externhost setting if it matches
rtptimeout=20                   ; Terminate call if 60 seconds of no RTP or RTCP activity
rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP or RTCP activity
;rtpkeepalive=<secs>            ; Send keepalives in the RTP stream to keep NAT open
;sipdebug = yes                 ; Turn on SIP debugging by default, from
;recordhistory=yes              ; Record SIP history by default
;dumphistory=yes                ; Dump SIP history at end of SIP dialogue
;allowsubscribe=no              ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default     ; Set a specific context for SUBSCRIBE requests
notifyringing = yes             ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes                ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes              ; Apply call limits on peers only. This will improve
;t38pt_udptl = yes            ; Default false
;register =>
;registertimeout=20             ; retry registration calls every 20 seconds (default)
;registerattempts=10            ; Number of registration attempts before we give up
;externip =        ; Address that we're going to put in outbound SIP
;     ; Alternatively you can specify a domain
;externrefresh=10               ; How often to refresh externhost if
;localnet=; All RFC 1918 addresses are local networks
;localnet=     ; Also RFC1918
;localnet=          ; Another RFC1918 with CIDR notation
;localnet= ;Zero conf local network
nat=force_rport,comedia         ; Global NAT settings  (Affects all peers and users)
canreinvite=no                  ; Asterisk by default tries to redirect the
;directrtpsetup=yes             ; Enable the new experimental direct RTP setup. This sets up
;rtcachefriends=yes             ; Cache realtime friends by adding them to the internal list
;rtsavesysname=yes              ; Save systemname in realtime database at registration
;rtupdate=yes                   ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes                ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes            ; Enabling this setting has two functions:
;domain=                 ; Add IP address as local domain
;allowexternaldomains=no        ; Disable INVITE and REFER to non-local domains
;autodomain=yes                 ; Turn this on to have Asterisk add local host
;fromdomain=mydomain.tld        ; When making outbound SIP INVITEs to
jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100             ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes             ; By default, qualify all peers at 2000ms
limitonpeer = yes       ; enable call limit on a per peer basis, different from limitonpeers
session-timers=refuse   ; Refuse WebRTC session timers

#include sip-vicidial.conf
#include sip-goautodial.conf

; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=
; dtmfmode=inband
; qualify=1000

It looks like you have the wrong dialplan entry along with possibly a bad caller id. look up trunk configurations for Go Autodial and see where you are wrong

Here is my dileplan

Any advice?

you apper to be using freePBX
you should therfor try to get help over at the freepbx forum, as we do not know how to configure freePBX
also please do not post screen shots please use the “Preformatted text” funktion

also please change from using chan_sip to use chan_pjsip as chan_sip is depricated

I don’t use Go AutoDial but you might want to check this link out and go with a much simpler dial plan than the one you have there.

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