[Asterisk-Users] Asterisk drops call when calling other VOIP

Tony Davidson tony at davidson.id.au
Thu Nov 17 15:01:32 MST 2005


Does anyone have any ideas for this??? 

Tony Davidson wrote:

> I don't think that's the issue as it works with 99.9% of people we call.
>
> It's only 2 numbers so far that have had this issue.  I'm pretty sure 
> one uses Asterisk at the other end, but I would have thought it 
> unlikely the second did.  I think it's one of those auto switch boards 
> though - maybe this is causing the issue?
>
> Tom Vile wrote:
>
>> its possible that your provider is not setup to use asterisk for your
>> account.  I know a some providers that need to know if you are using a
>> regular SIP phone or Asterisk.
>>
>> On 11/16/05, Tony Davidson <tony at davidson.id.au> wrote:
>>  
>>
>>> I'm having an issue when Asterisk calls what I believe to be other VOIP
>>> connections.
>>>
>>> I can call the number from a normal sip phone, but when I attempt to
>>> connect via Asterisk the call is dropped immediately.  Checking my call
>>> logs I can tell the call has connected but I think Asterisk is trying
>>> something when it connects that immediately causes a dropout.  My VOIP
>>> connection is via a SIP account.
>>>
>>> Tony
>>>
>>> The log of the call is:
>>>
>>>   -- Called engin/03XXXX0888
>>>    -- SIP/engin-f91f answered SIP/203-add5
>>>  == Spawn extension (macro-dialout-trunk, s, 17) exited non-zero on
>>> 'SIP/203-add5' in macro 'dialout-trunk'
>>>  == Spawn extension (from-internal, 0392210888, 1) exited non-zero on
>>> 'SIP/203-add5'
>>>    -- Executing Macro("SIP/203-add5", "hangupcall") in new stack
>>>    -- Executing ResetCDR("SIP/203-add5", "w") in new stack
>>>    -- Executing NoCDR("SIP/203-add5", "") in new stack
>>>    -- Executing Wait("SIP/203-add5", "5") in new stack
>>>    -- Executing Hangup("SIP/203-add5", "") in new stack
>>>  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on
>>> 'SIP/203-add5' in macro 'hangupcall'
>>>  == Spawn extension (from-internal, h, 1) exited non-zero on 
>>> 'SIP/203-add5'
>>> asterisk1*CLI>
>>>
>>>
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>>
>>
>>
>> -- 
>> Tom Vile
>> Baldwin Technology Solutions, Inc
>> Consulting - Web Design - VoIP Telephony
>> www.baldwintechsolutions.com
>> Phone: 518-631-2855 x205
>> Phone: 978-203-3848 x205
>> Fax:     518-631-2856
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com --
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>> Asterisk-Users mailing list
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>>
>>
>>  
>>
>
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