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 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
