[Asterisk-Dev] SIP Interworking

[Rob Gagnon]

For SIP specific code, you will want to look in the chan_sip.c file (inside
the "channels" directory), as well as the documentation in the sample
sip.conf file (inside the "configs" directory)

Inside chan_sip.c look for the function "load_module".. This is where the
initialization of the module begins when it is dynamically loaded when
asterisk starts.  I would say that all the asterisk .so (shared object)
files are initialized in the same manner, so the "load_module" function is
always a good place to start with any module.

For channel handlers, such as chan_sip.c, you should find a call to
ast_channel_register_ex() which registers the module as a channel handler,
and sets up pointer to call-back functions needed to handle a call....

 From there, you should have a good start at tracing the operation of the
code.

If you are simply trying to debug the SIP traffic between Asterisk and the
"cheap phones" then you might want to try just using the "sip debug" CLI
command while asterisk is running so you can see the actual SIP messaging
between the two.

Rob

----- Original Message ----- 
From: "jd dzeng" <jddzeng at hotmail.com>
To: <asterisk-dev at lists.digium.com>
Sent: Monday, July 26, 2004 7:42 PM
Subject: [Asterisk-Dev] SIP Interworking


> Hi,
>
> I got some SIP interworking problems when connecting Asterisk to some
cheap
> SIP phones.
> I want to know which source files are responsible for handling the SIP
> protocols, so that I can trace and modify the source codes if necessary.
> Also, is there any reference documents which describe the structure of the
> source codes that I can trace the source codes more efficiently?
>
> JD Dzeng
>
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