Outbound Dialling/Dial Plan issue (not working)

[jollos]

Asterisk 1.8.23.0-vici
centos 6.6
kernel: 2.6.32-504.23.4.el6.x86_64

Hi,

I have setup the majority of the system, but have not been able to get the outbound dialling working. I am testing via softphone and trying to dial out directly (without going through vicidial interface) I assume this is possible.

The latest dialplan is giving the sound of dialling out, but the destination is not ringing.

Any unnecessary x'outing is due to inability to post due to this message "Your post looks too spamy for a new user, please remove off-site URLs.", and then "Your post looks too spamy for a new user, please remove bad words or non-english text.

Your help will be greatly appreciated and thanks in advance!

Here is my config.

Registration string:
register => 0000000000000:6000000:0300000000_trunk@211.00.00.00/030000000000
Account Entry

[siptrunk]
disallow=all
allow=alaw
allow=G729
host=0000000000
type=friend
username=0300000000000_trunk
secret=6000000
insecure=port,invite
fromdomain=0000000000
canreinvite=yes
qualify=no
dtmfmode=auto
context=trunkinbound

Global String:
SIPstring= SIP/trunkinbound

exten =>_X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten =>_X.,2,Dial(${SIPstring}/${EXTEN:1}@siptrunk,,tTor)
exten =>_X.,3,Hangup

Globals:

[globals]
CONSOLE=Console/dsp ; Console interface for demo
;TRUNK=DAHDI/r1 ; Trunk interface
TRUNK=SIP/siptrunk
TRUNKX=DAHDI/r2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:0000000000 ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:00000000000 ; IAX trunk interface
TRUNKBINFONE=IAX2/1112223333:PASSWORD@000000000 ; IAX trunk interface
;SIPtrunk=000000000000 ; SIP trunk

Sip.conf

[general]
context=trunkinbound ; Default context for incoming calls
allowguest=yes ; 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

;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

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

;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

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Is there a reason you are installing on CentOS? Did you use Goautodial?

4) Can you just use the Vicibox.com installer?

5) The best/simplest/fastest method to get a functional Vicidial system is to install using the Vicibox.com iso image, with the PDF instructions available on that site, and as soon as you complete the installation you switch to The Vicidial Managers Manual for configuration, starting at page ONE and NOT skipping any pages (no matter how smart you are! LOL).

6) That being said, do not edit any of the configuration files in /etc/asterisk manually. All changes should be in the Vicidial interface. Please post an asterisk CLI output example of an attempted phone call, and we can see what happened.

7) When you post your "Carrier" settings from "Admin->Carriers", please be sure to include the field names before the values so we can be sure everything is in the right place. Misunderstandings in the technical world happen very easily when everyone "assumes" that the other person knows what they are doing and posted based on that knowledge we are sure they have, right?

In the future, please do not post your sip.conf file in its entirety. Two reasons: Waste of space and you should not be editing it anyway! Put the original file back where it was (there are extra copies based on the asterisk version you are using, the "locate" command will find it for you, although you may need to "updatedb" first to index the drive).

Happy Hunting!