Problem with stuttering dial tone using DISA

[quote]Apr 27 22:57:26 WARNING[746]: channel.c:2103 ast_indicate: Unable to handle indication 8 for 'SIP/MY_IP_ADDRESS-00113f68’
Apr 27 22:57:35 ERROR[744]: chan_sip.c:11364 sipsock_read: We could NOT get the channel lock for SIP/MY_IP_ADDRESS-00113f68!
Apr 27 22:57:35 ERROR[744]: chan_sip.c:11365 sipsock_read: SIP MESSAGE JUST IGNORED: BYE
Apr 27 22:57:35 ERROR[744]: chan_sip.c:11366 sipsock_read: BAD! BAD! BAD![/quote]
I want to use DISA so I can dial in to my Asterisk PBX from outside and get a dial tone so that I can dial out long distance at my VoIP rates. The way it works is I authenticate with a passcode and then the system transfers me to a dial tone, from which I can dial out.

The problem is I haven’t done this before, so I’m not sure what’s wrong. Everything works fine up to the point where I’m supposed to get a dial tone so I can dial out. The dial tone sounds like it’s stuttering… I’m not sure how else to describe it.

The bolded error in the quoted text above is the one giving me concern. The bottom three only show up once in a while, but may be related. The bolded error shows up after about ~5sec of waiting on the line, with the stuttering dial tone. If I hang up sooner than that, I get no error. If I try to enter any DTMF tones, nothing seems to happen.

I’m pretty new to Asterisk so I haven’t got too much idea on how to modify the many, many settings available. Any idea what could possibly resolve this, or what is happening? All other functions seem to work OK, without errors (ie: dialing out, voicemail, etc).

So it seems the last three errors (quoted above) are just problems with my sip.conf or extensions.conf … I am going to look through the files to see what I might have done wrong.

In the meantime, any ideas what could be wrong with my DISA set up?


any ideas?