[asterisk-users] Problem Hangup

[Knud Müller]

FERNANDO VILLARROEL schrieb:
> --- Knud Müller <k.mueller at portrix.net> wrote:
>
>   
>> FERNANDO VILLARROEL schrieb:
>>     
>>> Hello list, i need help.
>>>
>>> My problem is that when I want to call (using ISDN
>>> phone or internal SIP client) via the Sip provider
>>>       
>> a
>>     
>>> real phone number (ISDN phone or internal SIP
>>>
>>>         Asterisk >> SIP ), I get a ring tone. When
>>>       
>> I
>>     
>>> now decide to hang up (e.g. if 
>>>
>>> nobody answers), the called telephone continues to
>>> ring almost forever.
>>>
>>> the sip.conf:
>>>
>>> [2563105]
>>> accountcode = 2563105
>>> username = 2563105
>>> secret = 135
>>> callerid = 412563105
>>> context = test
>>> canreinvite = no
>>> dtmfmode = rfc2833
>>> host = dynamic
>>> insecure = very
>>> language = es
>>> nat = yes
>>> qualify = yes
>>> type = friend
>>> disallow=all
>>> allow=g729
>>>
>>> [nyphone]
>>> accountcode=nyphone
>>> canreinvite=no
>>> reinvite=yes
>>> username=test770
>>> secret=test770
>>> dtmfmode=rfc2833
>>> host=72.55.143.XXX
>>> insecure=very
>>> language=es
>>> nat=no
>>> qualify=no
>>> type=peer
>>> disallow=all
>>> allow=g729
>>>
>>> I attach sip debug one call.
>>>
>>> I use Asterisk 1.2.13
>>>
>>> I hope you understand me and help.
>>>
>>> Best regards
>>>
>>> Fernando Villarroel Noriel.
>>> Chillan
>>> Chile
>>>
>>> Sorry my English.
>>>
>>> 			
>>> 	
>>> 			
>>> 	
>>> 		
>>> 	
>>> 	
>>> 		
>>> 	
>>>
>>>
>>>
>>>        
>>>
>>>       
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>   
>> If I got it right: you register to your SIP Provider
>> which provides a 
>> PSTN Number to you. You dial the PSTN Number which
>> is forwarded to your 
>> asterisk. Your asterisk dials the SIP phone
>> (nyphone)?
>>     
>
> Yes nyphone is my provider for everyone calls
> internationational (prefix 00)
>
> 2563105 is one number provided for my Telco (E1) and
> is one SIP client.
>   
So its an outbound call not an inbound call!
>   
>> Could you attach your dialplan?
>>     
>
> exten => _00X.,1,dial(sip/${EXTEN:2}@nyphone,45)
> exten => _00X.,2,hangup
>   
This looks OK. I'd recommend to record the SIP communication with your 
provider. Do on the CLI: "SIP debug". You should see after latest after 
45 seconds that asterisk sends an hangup request to your provider. If * 
sends the request it must be your provider. If it is not sent, something 
is wrong with your *.
>
> the called telephone continues to ring almost forever.
>
>   
>> Knud
>>
>> _______________________________________________
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>> http://www.api-digital.com--
>>
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>>   
>>
>>     
> http://lists.digium.com/mailman/listinfo/asterisk-users
>   
>
>
>
>        
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>   


-- 
Knud A. Müller
Geschäftsführer
Tel.: 040/398053-11
Fax: 040/398053-29
e-Mail: k.mueller at portrix.net

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