[asterisk-users] Vitelity Asterisk configuration help

Darren Severino darren.severino at celebrationtechnology.com
Mon Oct 6 16:49:49 CDT 2008


Stephen,   What exactly are you trying to accomplish? If you want basic call
in/out you're just about there. Changes need to be made in your
extensions.conf. Your phones, by default, are in the [default] context. In
other words when making a call it looks for extensions here. To allow
outbound calling include your outgoing context within the default. To
include it, at the bottom of the default context add "include => outgoing"
either of these should allow outgoing calling. As for incoming, add a Goto
as follows.

[inbound]
exten => 9045622082,1,Answer
exten => 9045622082,n,Goto(default,101,1)

That equates to "goto the default context, extension 101, at the 1st
priority" which is your Dial command.

Best Regards,Darren Severino


On Sat, Oct 4, 2008 at 1:30 PM, Stephen Reese <rsreese at gmail.com> wrote:

> I have a Asterisk server setup and I am able to connect to the server
> using a soft client 'x-lite' and call and leave a message on my second
> extension 102. I have setup a Vitelity account and add what I believe
> to be the correct information to my sip.conf and extension.conf. I
> would like to setup incoming and outgoing calls with voicemail
> support. I've searched all over but many of the full configurations
> that are available are a bit complex. Any tips or recommendations to
> get up and running would be great.
>
> sip.conf
> Code:
>
> [general]
> register => rsreese:pass at inbound18.vitelity.net:5060
> context=default                 ; Default context for incoming calls
> realm=ns1.neocipher.net         ; 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=neocipher.net            ; Set default domain for this host
> [101]
> type=friend ; allows incoming and outgoing calls
> username=101
> secret=test81
> mailbox=101
> callerid="Stephen" <101>
> host=dynamic
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> disallow=all
> allow=gsm
> [102]
> type=friend ; allows incoming and outgoing calls
> username=102
> secret=test81
> mailbox=102
> callerid=("Bob" <101>)
> host=dynamic
> dtmfmode=rfc2833
> canreinvite=yes
> allowguest=yes
> insecure=very
> promiscredir=yes
> musicclass=default              ; Sets the default music on hold class
> for all SIP calls
> [authentication]
> [vitel-inbound] ;(exact format/casing required)
> type=friend
> host=inbound18.vitelity.net
> context=inbound ;(ext-did or from-trunk for A at H)
> username=rsreese
> secret=pass
> allow=all
> insecure=very
> canreinvite=no
> [vitel-outbound] ;(exact format/casing required)
> type=friend
> host=outbound.vitelity.net
> context=inbound ;(ext-did or from-trunk for A at H)
> username=rsreese
> fromuser=rsreese
> trustrpid=yes
> sendrpid=yes
> secret=pass
> allow=all
> canreinvite=no
>
>
> extensions.conf
> Code:
>
> [general]
> static=yes
> writeprotect=yes
>
> [globals]
>
> [default]
>
> exten => 101,1,Dial(SIP/101,20)
> exten => 101,2,Voicemail(102)
>
> exten => 102,1,Dial(SIP/102,20)
> exten => 102,2,Voicemail(102)
>
> exten=>*98,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT})   ;This
> automatically calls the right mailbox using the ${CALLERIDNUM}
> variable in the current context (var ${CONTEXT}).
>
> [outgoing]
> exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@vitel-outbound)
> exten => _011.,1,Dial(SIP/${EXTEN}@vitel-outbound)
>
> exten => _911,1,Dial(SIP/911 at vitel-outbound)
>
> [inbound]
> exten => 9045622082,1,Answer
>
>
> voicemail.conf
> Code:
>
> [general]
> format=wav49|gsm|wav
> serveremail=asterisk
> attach=yes
> skipms=3000
> maxsilence=10
> silencethreshold=128
> maxlogins=3
> emaildateformat=%A, %B %d, %Y at %r
> sendvoicemail=yes       ; Context to Send voicemail from [option 5
> from the advanced menu]
> [zonemessages]
> eastern=America/New_York|'vm-received' Q 'digits/at' IMp
> central=America/Chicago|'vm-received' Q 'digits/at' IMp
> central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
> military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
> [default]
> 101 => 123,Stephen Rese,rsreese at gmail.com
> 102 => 123,Bob Dole,rsreese at gmail.com
> 1234 => 4242,Example Mailbox,root at localhost
> [other]
> 1234 => 5678,Company2 User,root at localhost
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081006/fe4d6943/attachment-0001.htm 


More information about the asterisk-users mailing list