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

Jonas Arndt jonas_arndt at comcast.net
Tue Oct 16 22:26:30 CDT 2007



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



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