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Pablo Mora

2006-Jul-27 18:18 UTC

[asterisk-users] Strange behaviour Panasonic KX-TD1232

Hello,

 

I've got asterisk running and almost working with Panasonic KX-TD1232

I said almost, because there's a strange behaviour when I make calls.

 

 -------              -----------------              ---------------------
-----------

| SIP | ---------- | ASTERISK | ---------- | PANASONIC | ------------ | PSTN
|

 -------              -----------------              ---------------------
----------

                                                           |         | 

                                                       -------     -------

                                                      | Ext1|  | Ext2|

                                                       -------     -------

 

When I make a call from PSTN to SIP, the call goes on successfully.

When I make a call from SIP to PSTN, the call goes on successfully.

When I make a call from Ext1 or Ext2 to SIP, the call goes on successfully.

When I make a calla from SIP to Ext1 (Ext2. ExtN), the Sip phone keeps
ringing and user behind Ext1 doesn't hear anything.  

 

It seams appear like Asterisk doesn't detect the answer on Ext1

 

Is there any way to figure it out??

 

Thanks

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Pablo L. Arturi

2006-Jul-29 19:24 UTC

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[asterisk-users] Strange behaviour Panasonic KX-TD1232

Hello Pablo, I think you should decribe with details how are you routing the
call between the SIP device and the extensions.

Pablo
  ----- Original Message ----- 
  From: Pablo Mora 
  To: asterisk-users@lists.digium.com 
  Sent: Thursday, July 27, 2006 10:18 PM
  Subject: [asterisk-users] Strange behaviour Panasonic KX-TD1232


  Hello,

   

  I've got asterisk running and almost working with Panasonic KX-TD1232

  I said almost, because there's a strange behaviour when I make calls.

   

   -------              -----------------              ---------------------    
-----------

  | SIP | ---------- | ASTERISK | ---------- | PANASONIC | ------------ | PSTN |

   -------              -----------------              ---------------------    
----------

                                                             |         | 

                                                         -------     -------

                                                        | Ext1|  | Ext2|

                                                         -------     -------

   

  When I make a call from PSTN to SIP, the call goes on successfully.

  When I make a call from SIP to PSTN, the call goes on successfully.

  When I make a call from Ext1 or Ext2 to SIP, the call goes on successfully.

  When I make a calla from SIP to Ext1 (Ext2. ExtN), the Sip phone keeps ringing
and user behind Ext1 doesn't hear anything.

   

  It seams appear like Asterisk doesn't detect the answer on Ext1

   

  Is there any way to figure it out??

   

  Thanks



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C F

2006-Jul-29 19:54 UTC

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[asterisk-users] Strange behaviour Panasonic KX-TD1232

How is asterisk connected to the Panasonic KX-TD1232?

On 7/27/06, Pablo Mora <pablo@espoltel.net> wrote:>
>
>
>
> Hello,
>
>
>
> I've got asterisk running and almost working with Panasonic KX-TD1232
>
> I said almost, because there's a strange behaviour when I make calls.
>
>
>
>  -------              -----------------              ---------------------
>             -----------
>
> | SIP | ---------- | ASTERISK | ---------- | PANASONIC | ------------ |
PSTN
> |
>
>  -------              -----------------
> ---------------------                ----------
>
>
> |         |
>
>
> -------     -------
>
>                                                       |
> Ext1|  | Ext2|
>
>
> -------     -------
>
>
>
> When I make a call from PSTN to SIP, the call goes on successfully.
>
> When I make a call from SIP to PSTN, the call goes on successfully.
>
> When I make a call from Ext1 or Ext2 to SIP, the call goes on successfully.
>
> When I make a calla from SIP to Ext1 (Ext2? ExtN), the Sip phone keeps
> ringing and user behind Ext1 doesn't hear anything.
>
>
>
> It seams appear like Asterisk doesn't detect the answer on Ext1
>
>
>
> Is there any way to figure it out??
>
>
>
> Thanks
> _______________________________________________
> --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
>
>
>

Pablo Mora

2006-Aug-01 15:57 UTC

head link

[asterisk-users] Strange behaviour Panasonic KX-TD1232

Ok Ok, the figure doesn't help.
 
Here we go again.
 
 
 -----         ----------          -----------           ------
| SIP | ----- | ASTERISK | ------ | PANASONIC | ------- | PSTN |
 -----         ----------          -----------           ------
                                       |   |
                                    Ext1  Ext2
 
 
Here is my dialplan
 
[incoming]
exten => s,1,Answer
exten => s,2,Background(prueba-pbx)
exten => s,3,Set(TIMEOUT(response)=5)
exten => 1001,1,Dial,SIP/1001|20
exten => 1001,2,Hangup
exten => 1001,102,Congestion,3
exten => 1002,1,Dial,SIP/1002|20
exten => 1002,2,Hangup
exten => 1002,102,Congestion,3
 
[sip]
include => outgoing
exten => 1001,1,Dial(SIP/1001,20)
exten => 1001,2,Hangup
exten => 1001,102,Congestion,3
exten => 1002,1,Dial(SIP/1002,20)
exten => 1002,2,Hangup
exten => 1002,102,Congestion,3
 
[outgoing]
exten => 0,1,Dial,Zap/g1
exten => 0,2,Congestion
exten => 0,102,Congestion
 
exten => 9,1,Dial,Zap/g1/9
exten => 9,2,Congestion
exten => 9,102,Congestion
 
When I make a call from PSTN to SIP, first Answer the Panasonic, after this
I digit an Extension and the call goes to asterisk, then I dial to sip and
the call goes on successfully. 
When I make a call from Ext1 or Ext2 to SIP, I dial an extension and the
call goes to asterisk, then I dial to sip and the call goes on.
When I make a call from SIP to PSTN, first dial 9 and asterisk gets a Zap
sending 9 to get PSTN line, the dial the PSTN number and the call goes on.
When I make a call from SIP to Ext1 (Ext2. ExtN), the Sip phone keeps
ringing and user behind Ext1 doesn't hear anything.
 
Your help will be appreciated.
 
 
 
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Pablo Mora

2006-Aug-01 18:23 UTC

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[asterisk-users] Strange behaviour Panasonic KX-TD1232

Ok,

 

I'm going to stop pictures

 

I have a Digium 4 FXO Card in my asterisk, and connect to Panasonic through
2 extensions (configured in a pool)

 

This means when you dial 200 (example) in Panasonic, the call goes to
asterisk and it answers.

 

In this sense, the answer is yes. replacing asterisk by a conventional
phone, I can dial and the phone rings.

 

The only way in wich call doesn't work is from Sip to Panasonic Ext.

 

I really don't think the problem is asterisk, but ringing cadence and
ringback tones from Panasonic.

 

 

 

 

 

 

 

 

 

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Pablo Mora

2006-Aug-02 07:29 UTC

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[asterisk-users] Strange behaviour Panasonic KX-TD1232

I think still didn't explain me clearly.

 

The problem is when I dial 0, in this case the asterisk take Zap (connected
directly to ext 200 from Panasonic), Panasonic gives tone, dial another
extension (ie 100), the extension rings but when answer the phone asterisk
keeps ringing. it doesn't detect when you pick up the phone.

 

 

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Pablo Mora

2006-Aug-03 15:04 UTC

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[asterisk-users] Strange behaviour Panasonic KX-TD1232

CF,

 

Adding www after Dial doesn't solve the trouble.

 

I think we are talking the same but I don't express correctly.

 

Did you saw my dialplan? I don't think I would have to add r.

 

Yes, I have installed a 4 FXO Card, with fxsks signalling. What I mean is I
understand FXO doesn't give the tone, but Panasonic.  

 

The cadence of ringing on Panasonic is a little different to the PSTN's
cadence, FXO detects properly PSTN cadence when a call goes to or come from
PSTN, and when a call goes from Panasonic to Asterisk, but doesn't make same
job with a call going from Asterisk to Panasonic. 

 

The Sip phone behind Asterisk make the call, keeps ringing until Panasonic
extension answer.. It's normal, but even Panasonic user pick up the phone
the Sip phone keeps ringing. to user on Sip pone, nobody answer his call,
however user on Panasonic pick up and doesn't hear anything.

 

I'm going crazy.

 

 

 

 

 

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Pablo Mora

2006-Aug-03 15:51 UTC

head link

[asterisk-users] Strange behaviour Panasonic KX-TD1232

Ok

 

 

Here goes dialplan 

 

[general]

static=yes

writeprotect=yes

 

[incoming]

exten => s,1,Answer

exten => s,2,Background(pbx)

exten => s,3,Set(TIMEOUT(response)=5)

exten => 1001,1,Dial,SIP/1001|20

exten => 1001,2,Hangup

exten => 1001,102,Congestion,3

exten => 1002,1,Dial,SIP/1002|20

exten => 1002,2,Hangup

exten => 1002,102,Congestion,3

 

[sip]

include => outgoing

exten => 1001,1,Dial(SIP/1001,20)

exten => 1001,2,Hangup

exten => 1001,102,Congestion,3

exten => 1002,1,Dial(SIP/1002,20)

exten => 1002,2,Hangup

exten => 1002,102,Congestion,3

 

[outgoing]

exten => 0,1,Dial,Zap/g1

exten => 0,2,Hangup

exten => 0,102,Congestion

 

exten => 9,1,Dial,Zap/g1/9

exten => 9,2,Hangup

exten => 9,102,Congestion

 

 

 

Here goes zapata

 

[trunkgroups]

 

[channels]

context=default

switchtype=national

signalling=fxs_ks

rxwink=300              

usecallerid=no

hidecallerid=no

callwaiting=no

callprogress=yes

;progzone=us

usecallingpres=yes

threewaycalling=no

transfer=no

cancallforward=yes

callreturn=yes

echocancel=yes

echocancelwhenbridged=yes

busydetect=yes

rxgain=0.0

txgain=0.0

 

group=1

callgroup=1

pickupgroup=1

immediate=no

signalling=fxs_ks

group=1

callerid=asreceived

context=incoming

channel =>1

channel =>2

channel =>3

channel =>4

 

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C F

2006-Aug-03 16:17 UTC

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[asterisk-users] Strange behaviour Panasonic KX-TD1232

Change:

callprogress=yes


To:
callprogress=no


Also when dialing over the Zap FXO ports make sure to add the ww
before the DTMF digits so that your extension.conf reads like this:

exten => 9,1,Dial(Zap/g1/ww9 )


On 8/3/06, Pablo Mora <pablo@espoltel.net> wrote:>
>
>
>
> Ok
>
>
>
>
>
> Here goes dialplan
>
>
>
> [general]
>
> static=yes
>
> writeprotect=yes
>
>
>
> [incoming]
>
> exten => s,1,Answer
>
> exten => s,2,Background(pbx)
>
> exten => s,3,Set(TIMEOUT(response)=5)
>
> exten => 1001,1,Dial,SIP/1001|20
>
> exten => 1001,2,Hangup
>
> exten => 1001,102,Congestion,3
>
> exten => 1002,1,Dial,SIP/1002|20
>
> exten => 1002,2,Hangup
>
> exten => 1002,102,Congestion,3
>
>
>
> [sip]
>
> include => outgoing
>
> exten => 1001,1,Dial(SIP/1001,20)
>
> exten => 1001,2,Hangup
>
> exten => 1001,102,Congestion,3
>
> exten => 1002,1,Dial(SIP/1002,20)
>
> exten => 1002,2,Hangup
>
> exten => 1002,102,Congestion,3
>
>
>
> [outgoing]
>
> exten => 0,1,Dial,Zap/g1
>
> exten => 0,2,Hangup
>
> exten => 0,102,Congestion
>
>
>
> exten => 9,1,Dial,Zap/g1/9
>
> exten => 9,2,Hangup
>
> exten => 9,102,Congestion
>
>
>
>
>
>
>
> Here goes zapata
>
>
>
> [trunkgroups]
>
>
>
> [channels]
>
> context=default
>
> switchtype=national
>
> signalling=fxs_ks
>
> rxwink=300
>
> usecallerid=no
>
> hidecallerid=no
>
> callwaiting=no
>
> callprogress=yes
>
> ;progzone=us
>
> usecallingpres=yes
>
> threewaycalling=no
>
> transfer=no
>
> cancallforward=yes
>
> callreturn=yes
>
> echocancel=yes
>
> echocancelwhenbridged=yes
>
> busydetect=yes
>
> rxgain=0.0
>
> txgain=0.0
>
>
>
> group=1
>
> callgroup=1
>
> pickupgroup=1
>
> immediate=no
>
> signalling=fxs_ks
>
> group=1
>
> callerid=asreceived
>
> context=incoming
>
> channel =>1
>
> channel =>2
>
> channel =>3
>
> channel =>4
>
>
> _______________________________________________
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>

Eric "ManxPower" Wieling

2006-Aug-03 20:52 UTC

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[asterisk-users] Strange behaviour Panasonic KX-TD1232

Pablo Mora wrote:> [outgoing]
> 
> exten => 0,1,Dial,Zap/g1
> 
> exten => 0,2,Hangup
> 
> exten => 0,102,Congestion
You NEVER want Dial,Zap/g1

You If you want to just get an outside dialtone you ALWAYS want a trailing /

Dial,Zap/g1/


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