Re: [asterisk-users] Vitelity Asterisk configuration help
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([EMAIL PROTECTED])
;exten = 101,102,Voicemail(102)
exten=*98,1,VoiceMailMain([EMAIL PROTECTED]) ;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 = _1NXXNXX,1,Set(CALLERID(num)=XX)
;exten = _1NXXNXX,2,Set(CALLERID(name)=XX)
; The following will display your number on a caller ID
exten = _1NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _1NXXNXX,n,Set(CALLERID(name)=9045622082)
exten = _1NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
;exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
;exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Dial(SIP/[EMAIL PROTECTED])
; e911 must be enabled. see DIDs NPANXXNXXX Action e911
exten = _911,1,Dial(SIP/[EMAIL PROTECTED])
[inbound]
exten = 9045622082,1,Goto(default,101,1)
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Re: [asterisk-users] Vitelity Asterisk configuration help
Are you dialing a 1 before every number? That is required unless you make another pattern match. exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) 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. ___ -- 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
Re: [asterisk-users] Vitelity Asterisk configuration help
The voicemail command should be Voicemail([EMAIL PROTECTED]) so in extensions.conf exten = 101,n,Voicemail([EMAIL PROTECTED]) As for the console when you launch it add v's to set the debugging level 'asterisk -vr' 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/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- 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? ___ -- 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
Re: [asterisk-users] Vitelity Asterisk configuration help
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. On Tue, Oct 7, 2008 at 10:38 AM, Stephen Reese [EMAIL PROTECTED] wrote: The voicemail command should be Voicemail([EMAIL PROTECTED]) so in extensions.conf exten = 101,n,Voicemail([EMAIL PROTECTED]) As for the console when you launch it add v's to set the debugging level 'asterisk -vr' 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/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- 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? ___ -- 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
Re: [asterisk-users] Vitelity Asterisk configuration help
Darren Severino wrote: 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. You might also want to just check your settings at Vitelity. Over the last six months they have changed the server I'm support to connect to two or three times so my * box was not connecting to them. Therefor no service. I've I'd had it up for more than testing, and been testing, I'd have notices if there was any rime or reason for the changes. No notifications even. Rod -- On Tue, Oct 7, 2008 at 10:38 AM, Stephen Reese [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The voicemail command should be Voicemail([EMAIL PROTECTED]) so in extensions.conf exten = 101,n,Voicemail([EMAIL PROTECTED]) As for the console when you launch it add v's to set the debugging level 'asterisk -vr' 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/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- 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? ___ -- 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 ___ -- 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
Re: [asterisk-users] Vitelity Asterisk configuration help
Interesting, I've been using them since April and haven't had a problem. I know they changed their server settings a while back but didn't notice anything recently. On Tue, Oct 7, 2008 at 11:47 AM, Roderick A. Anderson [EMAIL PROTECTED]wrote: Darren Severino wrote: 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. You might also want to just check your settings at Vitelity. Over the last six months they have changed the server I'm support to connect to two or three times so my * box was not connecting to them. Therefor no service. I've I'd had it up for more than testing, and been testing, I'd have notices if there was any rime or reason for the changes. No notifications even. Rod -- On Tue, Oct 7, 2008 at 10:38 AM, Stephen Reese [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The voicemail command should be Voicemail([EMAIL PROTECTED]) so in extensions.conf exten = 101,n,Voicemail([EMAIL PROTECTED]) As for the console when you launch it add v's to set the debugging level 'asterisk -vr' 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/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- 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? ___ -- 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 ___ -- 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 ___ -- 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
Re: [asterisk-users] Vitelity Asterisk configuration help
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 [EMAIL PROTECTED] 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:[EMAIL PROTECTED]: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 [EMAIL PROTECTED])
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 [EMAIL PROTECTED])
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([EMAIL PROTECTED]) ;This
automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
[outgoing]
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Dial(SIP/[EMAIL PROTECTED])
[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,[EMAIL PROTECTED]
102 = 123,Bob Dole,[EMAIL PROTECTED]
1234 = 4242,Example Mailbox,[EMAIL PROTECTED]
[other]
1234 = 5678,Company2 User,[EMAIL PROTECTED]
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Re: [asterisk-users] Vitelity Asterisk configuration help
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([EMAIL PROTECTED]) ;This automatically
calls the right mailbox using the ${CALLERIDNUM} variable in the current
context (var ${CONTEXT}).
include = outgoing
include = inbound
[outgoing]
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Dial(SIP/[EMAIL PROTECTED])
; e911 must be enabled. see DIDs NPANXXNXXX Action e911
exten = _911,1,Dial(SIP/[EMAIL PROTECTED])
[inbound]
exten = 9045622082,1,Goto(default,101,1)
Sip.conf
[general]
register = rsreese:[EMAIL PROTECTED]: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 [EMAIL PROTECTED])
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 [EMAIL PROTECTED])
username=rsreese
fromuser=rsreese
trustrpid=yes
sendrpid=yes
secret=key
allow=all
canreinvite=no
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[asterisk-users] 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:[EMAIL PROTECTED]: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 [EMAIL PROTECTED])
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 [EMAIL PROTECTED])
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([EMAIL PROTECTED]) ;This
automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
[outgoing]
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Dial(SIP/[EMAIL PROTECTED])
[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,[EMAIL PROTECTED]
102 = 123,Bob Dole,[EMAIL PROTECTED]
1234 = 4242,Example Mailbox,[EMAIL PROTECTED]
[other]
1234 = 5678,Company2 User,[EMAIL PROTECTED]
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