[Asterisk-Users] SIP / ATA Issue was: Re: Voice-mail App

[Shawn L. Djernes]

Hello, 

I saw you talking about the REFER code but I have noticed something while 
running the the debug in my ATA186 

When trying to do a call transfer I get an XFER to the number and not a 
REFER.  Asterisk shows nothing on the sip debug messages but the BYE when 
the ATA thinks it has transfered the call. 

Anyone know if the ATA186 can be set to use REFER instead of XFER on 
transfer. 

Thansk,
Shawn 

James Sizemore writes: 

> In case you have never setup a 7960 before.  The easies
> way is to setup dhcp and have the code on a tftp server.
> option tftp-server-name         "your.tftp.edu"; 
> 
> If you need a copy of the xml configs or dial plans for the
> phone let me know, and I send some your way. 
> 
> There is a nasty bug with REFER in the SIP code
> for that phone before version P0S3-04-3-00.bin 
> 
> 
> Mark Spencer wrote: 
> 
>>> I have played with the timeouts :int timeout, int ftimeout
>>> for ast_readstring(chan, password, sizeof(password) - 1, 2000, 10000, 
>>> "#")
>>> To no effect, Could you give me a pointer to where
>>> I can start looking next to track down this strangeness.
>>> Maybe something in  the SIP driver?  Getting
>>> in and out of band DTMF?  etc...
>>>     
>>> 
>> 
>> Best place to look is probably the RTP code, where the digits are
>> generated.  Specifically look at the rfc2833 routines, assuming that's 
>> how
>> they're being sent.  I just got a 7960 on loan, so I'm going to set it up
>> so that I can try to duplicate any problems you're having. 
>> 
>> Mark 
>> 
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>>   
>> 
>  
> 
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