[asterisk-dev] astobj2 and chan_sip; first results... wanna test drive it?

Klaus Darilion klaus.mailinglists at pernau.at
Mon Jan 7 05:35:58 CST 2008

Steve Murphy schrieb:
> On Fri, 2008-01-04 at 09:32 +0100, Johansson Olle E wrote:
>> Steve,
>> This is work that has been on my wishlist for a very long time. Thank  
>> you!
>> I hope to get time to test drive this soon and have some ideas on how  
>> to do it.
>> We need to discuss the dialog matching to se fix it once and for all,  
>> there's a lot
>> of open issues I have there on my list from SIPit testings, some tests  
>> I could never
>> participate in because Asterisk crashed totally nad did everything  
>> wrong in the
>> matching...
>> Right now I'm busy finding new projects to work with on a consultancy  
>> basis
>> and a new sponsor for my Asterisk work.
>> /O
> Olle--
> I wish you absolute best of luck in your search for funding. I'm
> painfully aware of how stressful such efforts can be.
> I'd love to discuss some of the options with you, if you get a free
> moment. Until then, I have a few questions:
> 1. on dialog matching-- if headers and tags are to be used for matching
> multiple dialogs with the same callid, how often would this happen in
> the course of "normal" activity, and how many such dialogs (with the
> same callid) would normally exist; how many **COULD** exist? I'm going
> to try to read the sip spec so I can understand the whole dialog
> "thing". I'm not looking forward to it.

A really common scenario is Asterisk as PSTN gateway and (open)ser as 
SIP Proxy. A user is registered with 2 sip clients using one SIP account 
- thus the SIP proxy performs parallel forking. If there is an call from 
PSTN to SIP, Asterisk has an outgoing SIP call. This SIP call will be 
forked to 2 SIP clients by the SIP proxy. Thus, Asterisk will see 
responses from different callees: Same call-id but different to-tag. 
Usually this will create multiple "early" dialogs but only a single 
"confirmed" dialog. But in a race condition (both phones are picked up 
at the same time) it may happen that Asterisk gets 2 200ok responses. In 
this case Asterisk has to send BYE to all 200 ok responses except the 
first one.


> 2. utf-8 in asterisk-- It occurs to me that we switched asterisk to vi
> mode instead of emacs mode, utf-8 via the CLI might be easier. The emacs
> mode insists on using that high-order bit to indicate stuff like M-x,
> etc. The guy who generated libedit never responded to my email; he may
> be dead/too busy/at a different email address/etc.
> 3. on sip destruction-- I have a crash in check_rtp_timeout, and it's
> one of those chicken-egg issues; the channel pointer, pvt->owner goes
> null while the routine is running. I need to either pull it into pieces
> or lock things or... perhaps you might have some advice.
> murf
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