Asterisk audio issue!


i installed Vicidial on CentOS Linux release 7.2.1511 (Core),
everything’s OK but the autodial dont work,manual calls are ok bot autodial call dont pass to the agents,
i see in the asterisk cli the calls answered but non pass to the agent in ready status.

i try a carrier SIP and carrier IAX2,but is the same,manual calls on vicidial are ok!the server is out Nat,is directly on public ip

I compiled vicidial manually, and I think I made any mistakes in the steps, but I tried everything but it is not working …

It might be a firewall issue? even if it 'stupid to think because I disabled …

ANSWER 119 62.3% 24 45.3% 12 37.5% 7 31.8% 2 50.0% 0 0.0%
BUSY 5 2.6% 1 1.9% 1 3.1% 1 4.5% 0 0.0% 0 0.0%
CANCEL 54 28.3% 22 41.5% 13 40.6% 9 40.9% 2 50.0% 0 0.0%
CHANUNAVAIL 13 6.8% 6 11.3% 6 18.8% 5 22.7% 0 0.0% 0 0.0%
TOTALS 191 53 32 22 4 0

Executing [8368@default:2] AGI(“Local/22239030361410@default-00000016;1”, “agi://”) in new stack
– AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=VODAFONE))
– AGI Script agi:// completed, returning 0
– Executing [8368@default:3] AGI(“Local/22239030361410@default-00000016;1”, “agi-VDAD_ALL_outbound.agi,NORMAL-----LB”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
– Executing [h@default:1] AGI(“Local/22239030361410@default-00000016;2”, “agi://–HVcauses–PRI-----NODEBUG-----16-----ANSWER-----23-----0”) in new stack
– AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
– Executing [8368@default:4] AGI(“IAX2/sipcc200-4629”, “agi-VDAD_ALL_outbound.agi,NORMAL-----LB”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
== Manager ‘sendcron’ logged on from
== Manager ‘sendcron’ logged off from
– AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0

my 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=5060 ; 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=60 ; 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= ; 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=no ; 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

#include sip-vicidial.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

there is something wrong?

please help my…
have any ide?

thank you


Autodial must be a function of Vicidial. Have you inquired on the Vicidial support sites?

I thought it was a sound problem, i post on the Vicidial forum too but still nobody answered…

This isn’t going to be an Asterisk problem, as evidenced by your dialplan execution snippet:

-- Executing [8368@default:3] AGI("Local/22239030361410@default-00000016;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi

Whatever the agi-VDAD_ALL_outbound.agi script does, it isn’t doing what you want (apparently). The only people who can help you with that are the authors of the AGI script.