asterisk users - Oct 2008 - Vitelity Asterisk configuration help

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

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
>
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> 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
Thanks I am now able to make incoming calls but I'm still unable to call
out. Notice anything else off.

extension.conf

[general]
 static=yes
 writeprotect=yes

[globals]

[default]

exten => 101,1,Dial(SIP/101,20)
exten => 101,2,Voicemail(102)
exten => 101,102,Voicemail(102)

exten=>*98,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT})   ;This automatically
calls the right mailbox using the ${CALLERIDNUM} variable in the current
context (var ${CONTEXT}).
include => outgoing
include => inbound

[outgoing]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@vitel-outbound)
exten => _011.,1,Dial(SIP/${EXTEN}@vitel-outbound)

; e911 must be enabled. see DIDs > NPANXXNXXX > Action > e911
exten => _911,1,Dial(SIP/911 at vitel-outbound)

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


Sip.conf

[general]
register => rsreese:key 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
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no

musicclass=default              ; Sets the default music on hold class for
all SIP calls
language=en                     ; Default language setting for all
users/peers

[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=key
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=key
allow=all
canreinvite=no

> Are you dialing a 1 before every number? That is required unless you make
> another pattern match.
> exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@vitel-outbound)
> Then it becomes 10-digit dialing without the need to dial a 1. If that
> doesn't work open up the asterisk console and attempt to make a call
and
> reply with any error messages.
I was not adding the 1 before the number but that didn't help. I
opened the console 'asterisk -r' but when attempting to call out
nothing happened. Is there some type of logging level that needs to be
turned up?

When I call in which does still work I do get the following errors and
of course voicemail doesn't work.:
Oct  7 09:38:08 WARNING[6146]: app_voicemail.c:2461 leave_voicemail:
No entry in voicemail config file for '102'
Oct  7 09:38:18 WARNING[6146]: pbx.c:2435 __ast_pbx_run: Timeout, but
no rule 't' in context 'default'

Thanks again for the help.

> The voicemail command should be Voicemail(extension at context) so in
> extensions.conf
> exten => 101,n,Voicemail(101 at default)
> As for the console when you launch it add v's to set the debugging
level
> 'asterisk -vvvvvr' you can also run 'core set debug X'
X=debug level 0-10 I
> believe. Just to make sure, you are doing a 'module reload' each
time you
> make changes to configuration files right?
Cool I've got voicemail :-). I am reloading it and have increased the
logging level.

When dialing out I'm seeing:

    -- Executing Dial("SIP/101-08183018",
"SIP/19046260705 at vitel-outbound") in new stack
    -- Called 19046270705 at vitel-outbound
    -- SIP/vitel-outbound-0818b178 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
Oct  7 10:34:34 WARNING[6465]: pbx.c:2435 __ast_pbx_run: Timeout, but
no rule 't' in context 'default'

Think it's a problem with vitelity?

> Well, after very quickly making a test call it's not Vitelity. It could
be
> something with your account? Might want to try opening a support ticket. If
> you want, create a sub account and e-mail me off list the username and
> password and I'll test it with my box or vice versa.
I am now able to make outgoing calls after much deliberation. I had to
add callerid to my outgoing...

Here's the extensions.conf

[general]
 static=yes
 writeprotect=yes

[globals]

[default]

exten => 101,1,Dial(SIP/101,20)
exten => 101,n,Voicemail(101 at default)
;exten => 101,102,Voicemail(102)

exten=>*98,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT})   ;This
automatically calls the right mailbox using the ${CALLERIDNUM} variable
in the current context (var ${CONTEXT}).
include => outgoing
include => inbound

[outgoing]
; The following gives an Unknown Caller ID
;exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=XXXXXXXXXX)
;exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=XXXXXXXXXX)

; The following will display your number on a caller ID
exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=9045622082)
exten => _1NXXNXXXXXX,n,Set(CALLERID(name)=9045622082)
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@vitel-outbound)
;exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@vitel-outbound)
;exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@vitel-outbound)
exten => _011.,1,Dial(SIP/${EXTEN}@vitel-outbound)

; e911 must be enabled. see DIDs > NPANXXNXXX > Action > e911
exten => _911,1,Dial(SIP/911 at vitel-outbound)

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