[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.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

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.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

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 )?

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

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 asterisk_l...@earthshod.co.ukwrote:

 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.

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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


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,)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

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 


-- 
Ben Franklin quote: 

Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety. 
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

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

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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