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Nora Lavelle

2006-Jan-26 18:07 UTC

[Asterisk-Users] extension to extension dialing

Sorry for all the newbie questions. I really appreciate everyone's help
today. 

 

Okay I've got outgoing and incoming calls working with no echo. yay! Now
I'm having an issue with SIP extension to extension calling. Any time I
dial another extension it goes right into voice mail.  My
extensions.conf is pretty small and rough but, here's what I have right
now. Most of it was taken from the voip-info website. Any help as always
VERY appreciated. 

 

Thanks again!

Nora Lavelle

 

# cat extensions.conf 

[incoming]

exten => s,1,Answer();

exten => s,2,Background(ssn-greeting);

exten => *,1,Directory(default)

exten => 205,1,Wait(2)

exten => 205,2,Record(/tmp/asterisk-recording:gsm)

exten => 205,3,Wait(2)

exten => 205,4,Playback(/tmp/asterisk-recording)

exten => 205,5,Wait(2)

exten => 205,6,Hangup 

 

[internal]

exten => 101,1,Macro(stdexten,SIP/101)

exten => 102,1,Macro(stdexten,SIP/102)

exten => 103,1,Macro(stdexten,SIP/103)

exten => 123,1,Macro(stdexten,SIP/123)

exten => 124,1,Macro(stdexten,SIP/124)

exten => 125,1,Macro(stdexten,SIP/125)

exten => 126,1,Macro(stdexten,SIP/126)

exten => 127,1,Macro(stdexten,SIP/127)

exten => 128,1,Macro(stdexten,SIP/128)

exten => 129,1,Macro(stdexten,SIP/129)

exten => 130,1,Macro(stdexten,SIP/130)

exten => 135,1,Macro(stdexten,SIP/135)

exten => 117,1,Macro(stdexten,SIP/117)

exten => 201,1,Macro(stdexten,SIP/201)

 

[voicemail]

exten => 300,1,Answer

exten => 300,2,VoicemailMain(ssn-voicemail-greeting)

exten => 300,3,Hangup

 

[local]

exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})

exten => _9NXXXXXX,2,Congestion

 

[longdistance]

exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})

exten => _91NXXNXXXXXX,2,Congestion

 

[macro-stdexten]

exten => s,1,Dial(${ARG1},20)

exten => s,2,Goto(s-${DIALSTATUS},1)

exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})

exten => s-NOANSWER,2,Goto(default,s,1)

exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})

exten => s-BUSY,2,Goto(default,s,1)

exten => s-.,1,Goto(s-NOANSWER,1)

exten => a,1,VoicemailMain(${MACRO_EXTEN})

 

[default]

include => incoming

include => internal

include => voicemail

include => local

include => longdistance

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Gary Richardson

2006-Jan-26 18:18 UTC

head link

[Asterisk-Users] extension to extension dialing

Check your error messages in you asterisk console. Perhaps your sip
secret or caller id is broken?

What type of phones are you using?

On 1/26/06, Nora Lavelle <nora@silverspringnet.com>
wrote:>
>
>
> Sorry for all the newbie questions. I really appreciate everyone's help
> today.
>
>
>
> Okay I've got outgoing and incoming calls working with no echo. yay!
Now I'm
> having an issue with SIP extension to extension calling. Any time I dial
> another extension it goes right into voice mail.  My extensions.conf is
> pretty small and rough but, here's what I have right now. Most of it
was
> taken from the voip-info website. Any help as always VERY appreciated.
>
>
>
> Thanks again!
>
> Nora Lavelle
>
>
>
> # cat extensions.conf
>
> [incoming]
>
> exten => s,1,Answer();
>
> exten => s,2,Background(ssn-greeting);
>
> exten => *,1,Directory(default)
>
> exten => 205,1,Wait(2)
>
> exten => 205,2,Record(/tmp/asterisk-recording:gsm)
>
> exten => 205,3,Wait(2)
>
> exten => 205,4,Playback(/tmp/asterisk-recording)
>
> exten => 205,5,Wait(2)
>
> exten => 205,6,Hangup
>
>
>
> [internal]
>
> exten => 101,1,Macro(stdexten,SIP/101)
>
> exten => 102,1,Macro(stdexten,SIP/102)
>
> exten => 103,1,Macro(stdexten,SIP/103)
>
> exten => 123,1,Macro(stdexten,SIP/123)
>
> exten => 124,1,Macro(stdexten,SIP/124)
>
> exten => 125,1,Macro(stdexten,SIP/125)
>
> exten => 126,1,Macro(stdexten,SIP/126)
>
> exten => 127,1,Macro(stdexten,SIP/127)
>
> exten => 128,1,Macro(stdexten,SIP/128)
>
> exten => 129,1,Macro(stdexten,SIP/129)
>
> exten => 130,1,Macro(stdexten,SIP/130)
>
> exten => 135,1,Macro(stdexten,SIP/135)
>
> exten => 117,1,Macro(stdexten,SIP/117)
>
> exten => 201,1,Macro(stdexten,SIP/201)
>
>
>
> [voicemail]
>
> exten => 300,1,Answer
>
> exten => 300,2,VoicemailMain(ssn-voicemail-greeting)
>
> exten => 300,3,Hangup
>
>
>
> [local]
>
> exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
>
> exten => _9NXXXXXX,2,Congestion
>
>
>
> [longdistance]
>
> exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
>
> exten => _91NXXNXXXXXX,2,Congestion
>
>
>
> [macro-stdexten]
>
> exten => s,1,Dial(${ARG1},20)
>
> exten => s,2,Goto(s-${DIALSTATUS},1)
>
> exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
>
> exten => s-NOANSWER,2,Goto(default,s,1)
>
> exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
>
> exten => s-BUSY,2,Goto(default,s,1)
>
> exten => s-.,1,Goto(s-NOANSWER,1)
>
> exten => a,1,VoicemailMain(${MACRO_EXTEN})
>
>
>
> [default]
>
> include => incoming
>
> include => internal
>
> include => voicemail
>
> include => local
>
> include => longdistance
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>

Nora Lavelle

2006-Jan-26 18:21 UTC

head link

[Asterisk-Users] extension to extension dialing

Here's what I get in the the log this is when extension 130 dials
extension 129. Thanks again ! 

nora  

    -- Executing Macro("SIP/130-a644", "stdexten|SIP/129")
in new stack
    -- Executing Dial("SIP/130-a644", "SIP/129|20") in new
stack
    -- Called 129
Jan 26 17:20:48 WARNING[28243]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call 4c6f199b2f02a5e05ff21c140eade0a1@10.200.1.234
for seqno 102 (Critical Request)
  == No one is available to answer at this time
    -- Executing Goto("SIP/130-a644", "s-NOANSWER|1") in new
stack
    -- Goto (macro-stdexten,s-NOANSWER,1)
    -- Executing VoiceMail("SIP/130-a644", "u129") in new
stack
    -- Playing 'voicemail/default/129/unavail' (language 'en')
  == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'SIP/130-a644' in macro 'stdexten'
  == Spawn extension (default, 129, 1) exited non-zero on 'SIP/130-a644'

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gary
Richardson
Sent: Thursday, January 26, 2006 5:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extension to extension dialing

Check your error messages in you asterisk console. Perhaps your sip
secret or caller id is broken?

What type of phones are you using?

On 1/26/06, Nora Lavelle <nora@silverspringnet.com>
wrote:>
>
>
> Sorry for all the newbie questions. I really appreciate everyone's
help> today.
>
>
>
> Okay I've got outgoing and incoming calls working with no echo. yay!
Now I'm> having an issue with SIP extension to extension calling. Any time I
dial> another extension it goes right into voice mail.  My extensions.conf
is> pretty small and rough but, here's what I have right now. Most of it
was> taken from the voip-info website. Any help as always VERY appreciated.
>
>
>
> Thanks again!
>
> Nora Lavelle
>
>
>
> # cat extensions.conf
>
> [incoming]
>
> exten => s,1,Answer();
>
> exten => s,2,Background(ssn-greeting);
>
> exten => *,1,Directory(default)
>
> exten => 205,1,Wait(2)
>
> exten => 205,2,Record(/tmp/asterisk-recording:gsm)
>
> exten => 205,3,Wait(2)
>
> exten => 205,4,Playback(/tmp/asterisk-recording)
>
> exten => 205,5,Wait(2)
>
> exten => 205,6,Hangup
>
>
>
> [internal]
>
> exten => 101,1,Macro(stdexten,SIP/101)
>
> exten => 102,1,Macro(stdexten,SIP/102)
>
> exten => 103,1,Macro(stdexten,SIP/103)
>
> exten => 123,1,Macro(stdexten,SIP/123)
>
> exten => 124,1,Macro(stdexten,SIP/124)
>
> exten => 125,1,Macro(stdexten,SIP/125)
>
> exten => 126,1,Macro(stdexten,SIP/126)
>
> exten => 127,1,Macro(stdexten,SIP/127)
>
> exten => 128,1,Macro(stdexten,SIP/128)
>
> exten => 129,1,Macro(stdexten,SIP/129)
>
> exten => 130,1,Macro(stdexten,SIP/130)
>
> exten => 135,1,Macro(stdexten,SIP/135)
>
> exten => 117,1,Macro(stdexten,SIP/117)
>
> exten => 201,1,Macro(stdexten,SIP/201)
>
>
>
> [voicemail]
>
> exten => 300,1,Answer
>
> exten => 300,2,VoicemailMain(ssn-voicemail-greeting)
>
> exten => 300,3,Hangup
>
>
>
> [local]
>
> exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
>
> exten => _9NXXXXXX,2,Congestion
>
>
>
> [longdistance]
>
> exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
>
> exten => _91NXXNXXXXXX,2,Congestion
>
>
>
> [macro-stdexten]
>
> exten => s,1,Dial(${ARG1},20)
>
> exten => s,2,Goto(s-${DIALSTATUS},1)
>
> exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
>
> exten => s-NOANSWER,2,Goto(default,s,1)
>
> exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
>
> exten => s-BUSY,2,Goto(default,s,1)
>
> exten => s-.,1,Goto(s-NOANSWER,1)
>
> exten => a,1,VoicemailMain(${MACRO_EXTEN})
>
>
>
> [default]
>
> include => incoming
>
> include => internal
>
> include => voicemail
>
> include => local
>
> include => longdistance
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Colin Anderson

2006-Jan-26 20:30 UTC

head link

[Asterisk-Users] extension to extension dialing

Snom's don't care, port 2057 is fine. Can you ping each phone from the
Linux
console?



-----Original Message-----
From: Gary Richardson [mailto:gary.richardson@gmail.com]
Sent: Thursday, January 26, 2006 7:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extension to extension dialing


In your sip.conf, make sure these phones have a Type=Friend entry and
a qualify=yes. I don't think the qualify=yes is required, but it helps
in debuging.

About the port, I'm not too sure about sipura and snom phones (I only
have Cisco phones :(). That could have something to do with it..

On 1/26/06, Nora Lavelle <nora@silverspringnet.com>
wrote:>
> Hi there gary. thanks so much for your help. we're using sipura-841 and
snom 320s.>
> Here's the sip show peers.. that's weird that extension 130 has
port
2057.. could that be the problem ?>
> -nora
>
> Name/username    Host            Dyn Nat ACL Mask             Port
Status>
> 201/201          10.200.0.56      D          255.255.255.255  5060
Unmonitor> ed
> 130/130          10.200.0.10      D          255.255.255.255  2057
Unmonitor> ed
> 129/129          10.200.0.5       D          255.255.255.255  5060
Unmonitor> ed
> 127/127          10.201.0.30      D          255.255.255.255  5060
Unmonitor> ed
> 126/126          10.201.0.29      D          255.255.255.255  5060
Unmonitor> ed
> 125/125          10.201.0.35      D          255.255.255.255  5060
Unmonitor> ed
> 124/124          10.201.0.31      D          255.255.255.255  5060
Unmonitor> ed
> 102/102          10.200.0.48      D          255.255.255.255  5060
Unmonitor> ed
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com on behalf of Gary Richardson
> Sent: Thu 1/26/2006 5:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] extension to extension dialing
>
> Check your error messages in you asterisk console. Perhaps your sip
> secret or caller id is broken?
>
> What type of phones are you using?
>
> On 1/26/06, Nora Lavelle <nora@silverspringnet.com> wrote:
> >
> >
> >
> > Sorry for all the newbie questions. I really appreciate everyone's
help
> > today.
> >
> >
> >
> > Okay I've got outgoing and incoming calls working with no echo.
yay! Now
I'm> > having an issue with SIP extension to extension calling. Any time I
dial
> > another extension it goes right into voice mail.  My extensions.conf
is
> > pretty small and rough but, here's what I have right now. Most of
it was
> > taken from the voip-info website. Any help as always VERY appreciated.
> >
> >
> >
> > Thanks again!
> >
> > Nora Lavelle
> >
> >
> >
> > # cat extensions.conf
> >
> > [incoming]
> >
> > exten => s,1,Answer();
> >
> > exten => s,2,Background(ssn-greeting);
> >
> > exten => *,1,Directory(default)
> >
> > exten => 205,1,Wait(2)
> >
> > exten => 205,2,Record(/tmp/asterisk-recording:gsm)
> >
> > exten => 205,3,Wait(2)
> >
> > exten => 205,4,Playback(/tmp/asterisk-recording)
> >
> > exten => 205,5,Wait(2)
> >
> > exten => 205,6,Hangup
> >
> >
> >
> > [internal]
> >
> > exten => 101,1,Macro(stdexten,SIP/101)
> >
> > exten => 102,1,Macro(stdexten,SIP/102)
> >
> > exten => 103,1,Macro(stdexten,SIP/103)
> >
> > exten => 123,1,Macro(stdexten,SIP/123)
> >
> > exten => 124,1,Macro(stdexten,SIP/124)
> >
> > exten => 125,1,Macro(stdexten,SIP/125)
> >
> > exten => 126,1,Macro(stdexten,SIP/126)
> >
> > exten => 127,1,Macro(stdexten,SIP/127)
> >
> > exten => 128,1,Macro(stdexten,SIP/128)
> >
> > exten => 129,1,Macro(stdexten,SIP/129)
> >
> > exten => 130,1,Macro(stdexten,SIP/130)
> >
> > exten => 135,1,Macro(stdexten,SIP/135)
> >
> > exten => 117,1,Macro(stdexten,SIP/117)
> >
> > exten => 201,1,Macro(stdexten,SIP/201)
> >
> >
> >
> > [voicemail]
> >
> > exten => 300,1,Answer
> >
> > exten => 300,2,VoicemailMain(ssn-voicemail-greeting)
> >
> > exten => 300,3,Hangup
> >
> >
> >
> > [local]
> >
> > exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
> >
> > exten => _9NXXXXXX,2,Congestion
> >
> >
> >
> > [longdistance]
> >
> > exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
> >
> > exten => _91NXXNXXXXXX,2,Congestion
> >
> >
> >
> > [macro-stdexten]
> >
> > exten => s,1,Dial(${ARG1},20)
> >
> > exten => s,2,Goto(s-${DIALSTATUS},1)
> >
> > exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
> >
> > exten => s-NOANSWER,2,Goto(default,s,1)
> >
> > exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
> >
> > exten => s-BUSY,2,Goto(default,s,1)
> >
> > exten => s-.,1,Goto(s-NOANSWER,1)
> >
> > exten => a,1,VoicemailMain(${MACRO_EXTEN})
> >
> >
> >
> > [default]
> >
> > include => incoming
> >
> > include => internal
> >
> > include => voicemail
> >
> > include => local
> >
> > include => longdistance
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Nora Lavelle

2006-Jan-27 11:28 UTC

head link

[Asterisk-Users] extension to extension dialing

Hmm.. I definitely have type=friend in the sip.conf and I added
qualify=yes but, I think that's default anyways.. When I call from the
outside and enter his extension it goes through to him fine but, when I
go extension to extension it automatically goes to voicemail.. Here are
the messages from the console:

    -- Executing Macro("SIP/130-58df", "stdexten|SIP/124")
in new stack
    -- Executing Dial("SIP/130-58df", "SIP/124|20") in new
stack
    -- Called 124
Jan 27 10:27:10 WARNING[28243]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call 7998a5e87b708f4374ca0ec212863b6d@10.200.1.234
for seqno 102 (Critical Request)
  == No one is available to answer at this time
    -- Executing Goto("SIP/130-58df", "s-NOANSWER|1") in new
stack
    -- Goto (macro-stdexten,s-NOANSWER,1)
    -- Executing VoiceMail("SIP/130-58df", "u124") in new
stack
    -- Playing 'voicemail/default/124/greet' (language 'en')
Jan 27 10:27:10 NOTICE[28243]: sched.c:290 ast_sched_del: Attempted to
delete non-existant schedule entry 22838!
    -- Playing 'vm-isunavail' (language 'en')
    -- Playing 'vm-intro' (language 'en')

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gary
Richardson
Sent: Thursday, January 26, 2006 6:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extension to extension dialing

In your sip.conf, make sure these phones have a Type=Friend entry and
a qualify=yes. I don't think the qualify=yes is required, but it helps
in debuging.

About the port, I'm not too sure about sipura and snom phones (I only
have Cisco phones :(). That could have something to do with it..

On 1/26/06, Nora Lavelle <nora@silverspringnet.com>
wrote:>
> Hi there gary. thanks so much for your help. we're using sipura-841
and snom 320s.>
> Here's the sip show peers.. that's weird that extension 130 has
port
2057.. could that be the problem ?>
> -nora
>
> Name/username    Host            Dyn Nat ACL Mask             Port
Status>
> 201/201          10.200.0.56      D          255.255.255.255  5060
Unmonitor> ed
> 130/130          10.200.0.10      D          255.255.255.255  2057
Unmonitor> ed
> 129/129          10.200.0.5       D          255.255.255.255  5060
Unmonitor> ed
> 127/127          10.201.0.30      D          255.255.255.255  5060
Unmonitor> ed
> 126/126          10.201.0.29      D          255.255.255.255  5060
Unmonitor> ed
> 125/125          10.201.0.35      D          255.255.255.255  5060
Unmonitor> ed
> 124/124          10.201.0.31      D          255.255.255.255  5060
Unmonitor> ed
> 102/102          10.200.0.48      D          255.255.255.255  5060
Unmonitor> ed
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com on behalf of Gary
Richardson> Sent: Thu 1/26/2006 5:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] extension to extension dialing
>
> Check your error messages in you asterisk console. Perhaps your sip
> secret or caller id is broken?
>
> What type of phones are you using?
>
> On 1/26/06, Nora Lavelle <nora@silverspringnet.com> wrote:
> >
> >
> >
> > Sorry for all the newbie questions. I really appreciate everyone's
help> > today.
> >
> >
> >
> > Okay I've got outgoing and incoming calls working with no echo.
yay!
Now I'm> > having an issue with SIP extension to extension calling. Any time I
dial> > another extension it goes right into voice mail.  My extensions.conf
is> > pretty small and rough but, here's what I have right now. Most of
it
was> > taken from the voip-info website. Any help as always VERY
appreciated.> >
> >
> >
> > Thanks again!
> >
> > Nora Lavelle
> >
> >
> >
> > # cat extensions.conf
> >
> > [incoming]
> >
> > exten => s,1,Answer();
> >
> > exten => s,2,Background(ssn-greeting);
> >
> > exten => *,1,Directory(default)
> >
> > exten => 205,1,Wait(2)
> >
> > exten => 205,2,Record(/tmp/asterisk-recording:gsm)
> >
> > exten => 205,3,Wait(2)
> >
> > exten => 205,4,Playback(/tmp/asterisk-recording)
> >
> > exten => 205,5,Wait(2)
> >
> > exten => 205,6,Hangup
> >
> >
> >
> > [internal]
> >
> > exten => 101,1,Macro(stdexten,SIP/101)
> >
> > exten => 102,1,Macro(stdexten,SIP/102)
> >
> > exten => 103,1,Macro(stdexten,SIP/103)
> >
> > exten => 123,1,Macro(stdexten,SIP/123)
> >
> > exten => 124,1,Macro(stdexten,SIP/124)
> >
> > exten => 125,1,Macro(stdexten,SIP/125)
> >
> > exten => 126,1,Macro(stdexten,SIP/126)
> >
> > exten => 127,1,Macro(stdexten,SIP/127)
> >
> > exten => 128,1,Macro(stdexten,SIP/128)
> >
> > exten => 129,1,Macro(stdexten,SIP/129)
> >
> > exten => 130,1,Macro(stdexten,SIP/130)
> >
> > exten => 135,1,Macro(stdexten,SIP/135)
> >
> > exten => 117,1,Macro(stdexten,SIP/117)
> >
> > exten => 201,1,Macro(stdexten,SIP/201)
> >
> >
> >
> > [voicemail]
> >
> > exten => 300,1,Answer
> >
> > exten => 300,2,VoicemailMain(ssn-voicemail-greeting)
> >
> > exten => 300,3,Hangup
> >
> >
> >
> > [local]
> >
> > exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
> >
> > exten => _9NXXXXXX,2,Congestion
> >
> >
> >
> > [longdistance]
> >
> > exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
> >
> > exten => _91NXXNXXXXXX,2,Congestion
> >
> >
> >
> > [macro-stdexten]
> >
> > exten => s,1,Dial(${ARG1},20)
> >
> > exten => s,2,Goto(s-${DIALSTATUS},1)
> >
> > exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
> >
> > exten => s-NOANSWER,2,Goto(default,s,1)
> >
> > exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
> >
> > exten => s-BUSY,2,Goto(default,s,1)
> >
> > exten => s-.,1,Goto(s-NOANSWER,1)
> >
> > exten => a,1,VoicemailMain(${MACRO_EXTEN})
> >
> >
> >
> > [default]
> >
> > include => incoming
> >
> > include => internal
> >
> > include => voicemail
> >
> > include => local
> >
> > include => longdistance
> > _______________________________________________
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> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
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>
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