Calls answered but does not pass to the agent

[donalddushi]

Hi,

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.

HANGUP STATUS 24 HOURS 6 HOURS 1 HOUR 15 MIN 5 MIN 1 MIN
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://127.0.0.1:4577/call_log") in new stack
-- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=VODAFONE))
-- <Local/22239030361410@default-00000016;1>AGI Script agi://127.0.0.1:4577/call_log 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://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----23-----0") in new stack
-- <IAX2/sipcc200-4629>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 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- <IAX2/sipcc200-4629>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0

please help my...
have any ide?

thank you

[mflorell]

Looks like a Local channel resolution issue, which means no audio on the call. Have you tried a different carrier?

[donalddushi]

yes 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 ...

[donalddushi]

my public ip is 95.10.24.29
the server is outside nat

my sip.conf

[general]
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=0.0.0.0 ; IP address to bind to (0.0.0.0 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
allow=gsm
mohinterpret=default
mohsuggest=default
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
;regcontext=sipregistrations
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=<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 => 1234:password@mysipprovider.com
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
externip = 95.10.24.29 ; Address that we're going to put in outbound SIP
;externhost=test.test.com ; Alternatively you can specify a domain
;externrefresh=10 ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;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=mydomain.tld,mydomain-incoming
;domain=1.2.3.4 ; 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@10.10.10.16:5060
;
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=inband
; qualify=1000


there is something wrong?

[williamconley]

1) never post your actual public IP. Always change something.

2) If you DO post your public IP, never turn off your firewall afterwards (for more than a minute or two).

3) If you DO post your public IP and turn off your firewall ... please note that Vicidial is not as secure as it could be.

I'll change your IP in your post now. LOL

4) Since this is a scratch installation, you should post your link to the instructions you used. When others have similar problems, it'll help searches bring the solutions together.

5) Always post your Vicidial Version with Build. (Always)

6) You can turn on sip debug, agi debug, and use the "screen -r asterisk" console to check for errors. You can also check the /var/log/astguiclient logs to see which process made a bad decision (perhaps). There are plenty of things that can go wrong in a scratch install.

7) WHY did you install from scratch instead of using the Vicibox.com iso installer?