[Asterisk-Users] RE: SIP i.e. Is something broken?

[WipeOut]

Dave Weis wrote:

>On Mon, 29 Sep 2003, Brian Capouch wrote:
>  
>
>>Christopher J. Wolff wrote:
>>    
>>
>>>Is it safe to assume that a fresh CVS build will not have the SIP
>>>translation problem described?
>>>
>>>Just FYI: I had similar problems for a while, and then I completely 
>>>scrapped my CVS directory and did a CVS CHECKOUT (instead of an update). 
>>>That solved the problem.
>>>      
>>>
>>I "checkout" rebuilt from CVS last night about 10:00 pm, and filed a bug 
>>report at that time.
>>It isn't *all* SIP calls for me, btw; my ATA186 works just fine, but my 
>>Budgetone won't work with the "broken" code. . .
>>It's as described in other mails--I can receive calls on the Budgetone 
>>but when I make them the RTP part is broken and the calls cut off the 
>>second they're set up.
>>    
>>
>
>I'm seeing the same thing with my budgetones and today's cvs. They are all 
>on the same network and worked on previous versions. I can call 
>voicemailmain and the console says that it is playing but I hear no sound. 
>Then it hangs up automatically.
>
>
>  
>
So the issue looks like it is to do with the Bugetone phones and the 
problem seems to have been introduced between Thurday last week and Sunday..

The reason I say Thursday last week is becasue I checked out a fresh 
copy of the CVS today with the -D "last Thursday" switch and its working 
with the Bugetones..

Anyone know what changed??