[asterisk-users] channel.c switches to gsm even when sip.conf only allows ulaw

[Jonas Arndt]

Jonas Arndt wrote:
> Kevin P. Fleming wrote:
>   
>> Jonas Arndt wrote:
>>
>>   
>>     
>>> It does this without caring about the fact that you are ONLY allowing
>>> ulaw in the channel configuration. I have so far played with SIP but it
>>> seems the behavior is there for other channels as well (briefly tried it
>>> on IAX as well)
>>>
>>> The problem with this is that some SIP providers (ViaTalk) only allows
>>> DTMF of the type inband, which only works on ulaw. Therefore this switch
>>> to GSM makes it impossible to enter the DISA or Authenticate password.
>>>     
>>>       
>> You are misunderstanding the message, and this why in general we ask
>> users who are not well versed in how Asterisk works to *not* enable the
>> DEBUG messages on the console.
>>
>> What you are seeing is *not* Asterisk changing the format of the audio
>> stream between it and the provider, but instead it is changing the
>> internal 'write format' of the channel to GSM (by putting a GSM to ULAW
>> transcoder in the path). It is doing this because you are running an
>> application that wants to play sound files, and you don't have ULAW
>> sound files installed. Since the only sound files you have are GSM, it
>> has to transcode that to ULAW before sending it out the channel.
>>
>>   
>>     
>
> Hi Kevin,
>
> Thanks for this. I appreciate the fact that I have a lot to learn as far
> as understanding the code in Asterisk and I am in fact looking forward
> to it. All pointers are very welcome.
>
> Let me state a couple of facts though:
> 1. After having upgraded to 1.4.12 this problem started. It is also
> there in 1.4.13
> 2. The SIP provider ViaTalk demands inband DTMF. Without it DTMF doesn't
> work
> 3. The initial IVR works and I can see how chan_sip.c detects DTMF
> 4. Once the caller has chosen something that leads them to an
> Authenticate (or DISA) in the dial plan the chan_SIP.c doesn't anymore
> detect DTMF.
> 5. This problem is not there if a SIP channel does not run inband DTMF
>
> So if you have any pointers as how to further troubleshoot this I would
> be very grateful. You are saying that I should not enable debug but
> without it I can only see the logic in the dial plan with Authenticate
> timing out. This logic has been working for over a year, so I don't
> really think that this is a problem.
>
> Thanks,
>
> // Jonas
>
>   

Hi Again,

I downgraded to asterisk 1.4.11 and the problem is still there. I
started tracing on dtmf and can see that the incoming DTMF are the
correct ones and I still get a "Password Incorrect" from the Authenticate.

THis is my simple conf in extensions.conf

[disa-custom]
exten => s,1,Wait(1)
exten => s,2,Authenticate(1234,,4)
exten => s,3,DISA(no-password,internal)

I am a bit confused at this point and I guess I have some more research
to do. I can't understand why it use to work and how/why it stopped.
Very strange....

Thanks,

// Jonas