Re: [asterisk-users] Incoming Call by DID

[A J Stiles]

On Wednesday 26 Oct 2016, KyD wrote:
> Hi,
> 
> My sip provider gave me 2 numbers for the incoming call via pstn.
> 
> nro1 = 12341234
> nro2 = 45674567
> 
> I have a dialplan for each.
> if i put this on my dialplan:
> 
> exten => s,1,Dial(SIP/1001)
> exten => Hangup()
> 
> Works!
> 
> But if i put one of them:
> 
> exten => 12341234,1,Dial(SIP/1001)
> exten => _1234,1,Dial(SIP/1001)
> exten => 45674567,1,Dial(SIP/1001)
> exten => _4567,1,Dial(SIP/1001)
> 
> incoming calls do not arrive.
> 
> Any ideas?

The incoming call must be arriving with ${EXTEN} containing something that 
doesn't match  12341234, _1234, 45674567 or _4567, so it is 
not triggering any of the extensions in your dialplan.  Maybe it still has the 
STD code or even the IDD code prepended.  (Been caught this way once before 
.  our old ISDN-30 provider used to send just the local number, then we 
moved to a new ISDN-30 provider who send the number with STD code but no 
initial 0.  Cue frantic editing of dialplan before rest of staff arrived .)
 
So try this;
 
exten => s,1,NoOp(Incoming call for '${EXTEN}')
exten => s,n,Dial(SIP/1001)
exten => s,n,Hangup()
 
Run `# asterisk -vvvr`, dial one of your DDI numbers from 
a mobile phone and watch the messages scrolling past.
 
Now you will be seeing exactly what ${EXTEN} contains when a call comes in, so 
you should be able to work out what is going on, and craft your extension 
expressions to suit.  If in doubt, post an excerpt from your CLI output.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] Incoming Call by DID

[David Duffett]

It seems like your SIP provider is not sending and DID information, or that
the information is not being sent in the same format you are using in your
dialplan.

You can check this by looking at the SIP debug information for the inbound
calls and/or by checking with your SIP provider (that they are sending the
DID number and what format it is in).

All the best,
David

On 27 Oct 2016 5:21 am, "KyD"  wrote:

Hi,

My sip provider gave me 2 numbers for the incoming call via pstn.

nro1 = 12341234
nro2 = 45674567

I have a dialplan for each.
if i put this on my dialplan:

exten => s,1,Dial(SIP/1001)
exten => Hangup()

Works!

But if i put one of them:

exten => 12341234,1,Dial(SIP/1001)
exten => _1234,1,Dial(SIP/1001)
exten => 45674567,1,Dial(SIP/1001)
exten => _4567,1,Dial(SIP/1001)

incoming calls do not arrive.

Any ideas?
--
KyD
GNU/Linux SysAdmin
Quanto mais você sabe, mais você percebe que você não sabe nada.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.
org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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 --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Incoming Call by DID

[KyD]

Hi,

My sip provider gave me 2 numbers for the incoming call via pstn.

nro1 = 12341234
nro2 = 45674567

I have a dialplan for each.
if i put this on my dialplan:

exten => s,1,Dial(SIP/1001)
exten => Hangup()

Works!

But if i put one of them:

exten => 12341234,1,Dial(SIP/1001)
exten => _1234,1,Dial(SIP/1001)
exten => 45674567,1,Dial(SIP/1001)
exten => _4567,1,Dial(SIP/1001)

incoming calls do not arrive.

Any ideas?
-- 
KyD
GNU/Linux SysAdmin
Quanto mais você sabe, mais você percebe que você não sabe nada.

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[Asterisk-Users] Incoming call and DID routing

[Rishabh Parikh]

Hello,



Just setup an asterisk server and Amp help me install their
portal and Asterisk. The server is up and all the hardware is
loading. Problem is with incoming calls. All calls go to the first
extension on the system and it still does not ring it goes straight to
voicemail. The person on ext. 5001 is busy please leave a message. 

Any ideas?? 



Thank you,



Rishabh






___
--Bandwidth and Colocation sponsored by Easynews.com --

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