[asterisk-users] This may be a problem.. Answer not working on 1.4.32 over SIP trunk..

[Massimo Nuvoli]

Zeeshan Zakaria ha scritto:
> I have two test asterisk boxes, both version 1.4.26, on which I do
> Answer() followed by MusicOnHold() and it works just fine. I do this all
> the time as this is my standard way of testing new contexts.

Yesterday i tested another installation and i found the same issue.

Maybe the problem is "SIP" related or "console channel" related.

I explain (if someone can do a test i am happy).

Go to the asterisk console, place a "dial" command calling thru the
SIP trunk, then place a "transfer" to the extension MusicOnHold after
the Answer...

(this is the sequence)

dial 0number at from-sip (the from-sip is the context where a sip phone
can dial to the trunk)
pick up the phone called
transfer *199 at from-sip (the *199 extension is "Answer -> MusicOnHold")
you must hear the music on the phone called (or not)

So this may be a "console channel problem"...

Yesterday i try to use the outgoing spool (place a file on
/var/spool/asterisk/outgoing making a call to the phone and directly
go to the *199 extension, the same thing i do on console automated
with no console channel), audio ok.

So i am going to open a bug... :-)

Thnks.

> Zeeshan A Zakaria
> 
> --
> www.ilovetovoip.com <http://www.ilovetovoip.com>
> 
>> On 2010-07-07 4:16 AM, "Massimo Nuvoli" <massimo at archivio.it
>> <mailto:massimo at archivio.it>> wrote:
>>
>> I found a strange thing on Asterisk 1.4.32, the same defect on 1.4.26.?
>>
>> I spend 4 hours to try to solve... but found only a workaround.
>>
>> As is easy to reproduce the problem i need to know if this is a bug or
>> if there is some idiot configuration that i miss.
>>
>> Maybe also the bug is know...
>>
>> Scenario:
>>
>> Asterisk installation on ubuntu 9.04 64 bit.
>>
>> Trunk SIP (two different providers)
>>
>> On the Asterisk server there are a number of SIP clients.
>>
>> If i use the sip client all things ok, i made a call and everything ok.
>>
>> If i place the call from the server (or if i call trhu the SIP trunk
>> the asterisk system) everytime the Answer() application seeems to NOT
>> work.
>>
>> The only way to make it work is to use some other function that do the
>> Answer in place.
>>
>> (call coming from the SIP trunk)
>> If i use
>>
>> Answer()
>> MusicOnHold()
>>
>> I hear nothing.
>>
>> If i use
>>
>> Answer()
>> PlayBack(silence/1)
>> MusicOnHold()
>>
>> or
>>
>> Answer()
>> VoiceMail(1234 at default)
>>
>> i can hear all ok (it seems that the PlayBack and the VoiceMail apps
>> are able to Answer really...)
>>
>> I checked the SIP debug trace, it seems no problem on the SIP side.
>>
>> Thnks guys.
>>
>> --
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