Re: [asterisk-users] extensions.conf / test DID

2013-04-08 Thread Steve Edwards 

On Mon, 8 Apr 2013, Thomas Perron wrote:

I am trying to make sure my DID and SIP account details are working 
properly and engaging the extensions.conf and dial plan.


If you jack up logging, you may see a message on the console like:

looking for x in y

where x is the extension and y is the context.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] extensions.conf / test DID

2013-04-08 Thread Doug Lytle 
>> I don't think s extension will work on SIP channel. s extension is a 
>> catch-all extension for Analog calls 

Console output would be useful. 

Doug 


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Re: [asterisk-users] extensions.conf / test DID

2013-04-08 Thread Satish Barot 
On Mon, Apr 8, 2013 at 4:26 PM, A J Stiles wrote:

> On Monday 08 April 2013, Thomas Perron wrote:
> > I am trying to make sure my DID and SIP account details are working
> > properly and engaging the extensions.conf and dial plan.
> >
> > I have a successful SIP session registered:
> >
> > Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
> > Asterisk*CLI> sip show registry
> > Hostdnsmgr Username   Refresh
> > StateReg.Time
> > sip3.voipvoip.com:5060  N  1112530146 105
> > Registered   Mon, 08 Apr 2013 06:02:09
> > 1 SIP registrations.
> > Asterisk*CLI>
> >
> > Here is the dial plan:
> > [incoming]
> > exten => 17036361355,1,Playback(beep)
> > exten => 17036361355,2,SayDigits(${EXTEN})
> > exten => 17036361355,3,Goto(testdtmf|s|1
> > ;Ring on Elle  mobile phone.
> > ;exten => s,1,Answer()
> > ;exten => s,n,Dial(SIP/17037171234,150,r,t,)
> >
> >
> > [general]
> > register =>1112530146:albany!@#1...@sip3.voipvoip.com/1112530146
> > registertimeout=20
> > context=incoming
> > allowoverlap=no
> > bindport=5060
> > bindaddr=192.168.1.10
> > srvlookup=no
> > ;context=incoming
> >
> > ; The SIP provider
> > [voipvoip.com]
> > canreinvite=no
> > username=1112530146
> > fromuser=1112530146
> > secret=albany!@#123
> > context=incoming
> > type=friend
> > fromdomain=s...@voipvoip.com
> > host=69.90.209.57
> > dtmfmode=rfc2833
> > disallow=all
> > allow=alaw
> > allow=ulaw
> > nat=force_rport
> > insecure=port,invite
> >
> > Thoughts please?I think something to do w/ "incoming" is incorrect.
>
> You only have one extension, "17036361355" in the [incoming] context in
> your
> dialplan.  Are you sure that "17036361355" is exactly what the SIP provider
> are actually sending to your end ?
>
> I'd put an "s" extension with a  NoOp(${EXTEN}) in there, just to catch the
> actual extension number they were sending.
>
> --
> AJS
>
> Answers come *after* questions.
>
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I don't think s extension will work on SIP channel. s extension is a
catch-all extension for Analog calls and Macros (reference:
https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions)

Just for the sake of testing I would have something like,
 [incoming]
 exten => _X.,1,NoOp(EXTENSION=${EXTEN})
 exten => _X.,2,Playback(beep)
 exten => _X.,3,SayDigits(${EXTEN})
 exten => _X.,3,Goto(testdtmf|s|1)
;Ring on Elle  mobile phone.
;exten => s,1,Answer()
 ;exten => s,n,Dial(SIP/17037171234,150,r,t,)
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Re: [asterisk-users] extensions.conf / test DID

2013-04-08 Thread Jacob . E . Miles 
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.

I have a successful SIP session registered:

Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI> sip show registry
Hostdnsmgr Username   Refresh
StateReg.Time
sip3.voipvoip.com:5060  N  1112530146 105
Registered   Mon, 08 Apr 2013 06:02:09
1 SIP registrations.
Asterisk*CLI>

Here is the dial plan:
[incoming]
exten => 17036361355,1,Playback(beep)
exten => 17036361355,2,SayDigits(${EXTEN})
exten => 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle  mobile phone.
;exten => s,1,Answer()
;exten => s,n,Dial(SIP/17037171234,150,r,t,)


[general]
register =>1112530146:albany!@#1...@sip3.voipvoip.com/1112530146
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.1.10
srvlookup=no
;context=incoming

; The SIP provider
[voipvoip.com]
canreinvite=no
username=1112530146
fromuser=1112530146
secret=albany!@#123
context=incoming
type=friend
fromdomain=s...@voipvoip.com
host=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=force_rport
insecure=port,invite

Thoughts please?I think something to do w/ "incoming" is incorrect.



 

[incoming]
exten => 17036361355,1,Playback(beep)
exten => 17036361355,2,SayDigits(${EXTEN})
exten => 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle  mobile phone.
;exten => s,1,Answer()
;exten => s,n,Dial(SIP/17037171234,150,r,t,)

 

As well doesn't the Goto need to closing ")"?

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Re: [asterisk-users] extensions.conf / test DID

2013-04-08 Thread A J Stiles 
On Monday 08 April 2013, Thomas Perron wrote:
> I am trying to make sure my DID and SIP account details are working
> properly and engaging the extensions.conf and dial plan.
> 
> I have a successful SIP session registered:
> 
> Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
> Asterisk*CLI> sip show registry
> Hostdnsmgr Username   Refresh
> StateReg.Time
> sip3.voipvoip.com:5060  N  1112530146 105
> Registered   Mon, 08 Apr 2013 06:02:09
> 1 SIP registrations.
> Asterisk*CLI>
> 
> Here is the dial plan:
> [incoming]
> exten => 17036361355,1,Playback(beep)
> exten => 17036361355,2,SayDigits(${EXTEN})
> exten => 17036361355,3,Goto(testdtmf|s|1
> ;Ring on Elle  mobile phone.
> ;exten => s,1,Answer()
> ;exten => s,n,Dial(SIP/17037171234,150,r,t,)
> 
> 
> [general]
> register =>1112530146:albany!@#1...@sip3.voipvoip.com/1112530146
> registertimeout=20
> context=incoming
> allowoverlap=no
> bindport=5060
> bindaddr=192.168.1.10
> srvlookup=no
> ;context=incoming
> 
> ; The SIP provider
> [voipvoip.com]
> canreinvite=no
> username=1112530146
> fromuser=1112530146
> secret=albany!@#123
> context=incoming
> type=friend
> fromdomain=s...@voipvoip.com
> host=69.90.209.57
> dtmfmode=rfc2833
> disallow=all
> allow=alaw
> allow=ulaw
> nat=force_rport
> insecure=port,invite
> 
> Thoughts please?I think something to do w/ "incoming" is incorrect.

You only have one extension, "17036361355" in the [incoming] context in your 
dialplan.  Are you sure that "17036361355" is exactly what the SIP provider 
are actually sending to your end ?

I'd put an "s" extension with a  NoOp(${EXTEN}) in there, just to catch the 
actual extension number they were sending.

-- 
AJS

Answers come *after* questions.

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[asterisk-users] extensions.conf / test DID

2013-04-08 Thread Thomas Perron 
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.

I have a successful SIP session registered:

Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI> sip show registry
Hostdnsmgr Username   Refresh
StateReg.Time
sip3.voipvoip.com:5060  N  1112530146 105
Registered   Mon, 08 Apr 2013 06:02:09
1 SIP registrations.
Asterisk*CLI>

Here is the dial plan:
[incoming]
exten => 17036361355,1,Playback(beep)
exten => 17036361355,2,SayDigits(${EXTEN})
exten => 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle  mobile phone.
;exten => s,1,Answer()
;exten => s,n,Dial(SIP/17037171234,150,r,t,)


[general]
register =>1112530146:albany!@#1...@sip3.voipvoip.com/1112530146
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.1.10
srvlookup=no
;context=incoming

; The SIP provider
[voipvoip.com]
canreinvite=no
username=1112530146
fromuser=1112530146
secret=albany!@#123
context=incoming
type=friend
fromdomain=s...@voipvoip.com
host=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=force_rport
insecure=port,invite

Thoughts please?I think something to do w/ "incoming" is incorrect.
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