[Shtoom] Re: [Divmod-dev] An application with your framework.

Anthony Baxter anthony at interlink.com.au
Fri Oct 1 11:33:14 CEST 2004

David Gilbert wrote:
> Well... I must start by saying that I'm impressed with your sip 
> framework.  I've got a few $'s to work on a project to product a SIP 
> thingie that calls a number, punches some DTMF and then records the 
> result (dialing into remote callcenter PBX's and monitoring a call for 
> later grading).  Anyways, I thought I'd let you know that I'm poking 
> along with your code.  It's a tough slog, but it's going.  So far I've 
> centered on using a bit of the annoucing message server as a model (A 
> 'doug' application) ... but it still seems to listen on 5060 by default 
> --- not exactly what I want.  Need to find that widget.

You might want to look at scripts/testcisco.py, as this does exactly
this - it connects to a host and interacts with an IVR. For applications
like this, passing -p0 (or --listenport=0) will "do the right thing" and
allocate a random port number.

> I've checked out the source with subversion and to the extent that I 
> modify things in your base classes, I'd like to contribute.  I also 
> don't know what your time/day is like for responding to email --- so if 
> it's tight, I'll just get back to you when I'm finished.  If not, I'd be 
> happy to sync up and keep an open dialogue.

My available time to spend on shtoom is highly variable, and depends on
a pile of other things. Either email, or the channel #shtoom on
irc.freenode.net, are the best way to contact me.

> One thing I've noticed off the top is that the shtoomphone.py 
> application seems to receive a stream of RTP, but not send.  I can hear 
> the "echo" of my input in my own speakers, but not in the phone I call 
> --- whereas taking in the called phone is audible in the speakers.  
> Using ethereal I determined that RTP packets were coming from the 
> provider but none were going to the provider (although the SIP 
> conversation appeared to be normal).

Dunno what's going on there - you might want to try using the
test harness (look in DEBUGGING.txt for details on how to run it) to
check that your audio devices are sane. In particular, if you're on
ALSA (a recent linux innovation), getting the settings right is a
completely black art. Hearing yourself talk, but not actually
generating any audio, often can mean a misconfiguration - if you
can hear yourself talk, that's often the problem. When running the
harness, you should only hear yourself talking when there's a call
up - if you can hear yourself otherwise, your sound system is
misconfigured :-(


Anthony Baxter     <anthony at interlink.com.au>
It's never too late to have a happy childhood.

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