[Asterisk-Dev] sip tester?

Nicholas Wong nwong at kancharla.com
Tue Jun 14 01:54:56 MST 2005


This will really help later with some other testing, but unfortunately I
need to test a response, and I'm pretty sure it will only work when it
is a response.

I've kinda been thrown into this pretty fast, I've only had a day to
look at this sip code.... but let me see if I got this right on a broad
scale:

phone makes a call:
persistant throughout call sip_pvt struct created and populated with info:
reads the dial plan to find the next hop
sends a 100 invite built from info in sip_pvt in a handle_request struct
the peer responds with "something" (or not) and it goes into sip_request
first,
has to be sip 2.0, pass a sanity test, be a response to
p->initreq.headers and be "in sequence"
Then the contact field has to have length and it is passed to
handle_response

And then yadda yadda yadda handle_response is switched off the sip xxx
number et. all.

I'm having to test changes to handle_request and handle_response.

Any kind of "I respond with this sip packet" daemon I can try? Or is
their a simpler way to get an exact response I need?

If you read this far, thanks.

Nick Wong


Darren Sessions wrote:

> We use it. Works great!
>
> - D
>
> On Jun 13, 2005, at 10:14 PM, Shanon Swafford wrote:
>
>>
>> Take a look at http://www.sipsak.org/
>>
>> Haven't tried it yet though.
>>
>> Shanon
>>
>> -----Original Message-----
>> From: asterisk-dev-bounces at lists.digium.com
>> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of  Nicholas
>> Wong
>> Sent: Monday, June 13, 2005 10:32 AM
>> To: Asterisk-Dev at lists.digium.com
>> Subject: [Asterisk-Dev] sip tester?
>>
>>
>> I'm working on some sip code, but I don't have any way to test it.
>>
>> Is there a debug app or something where I can send a sip message 
>> with the
>> headers I need to test?
>>
>> Thanks,
>>
>> Nick Wong
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