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

James Sizemore james at deny.org
Tue Feb 25 14:43:37 MST 2003


The ATAs do not yet use REFER to transfer they use the older XFER....So 
this is normal.


Shawn L. Djernes wrote:

> 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
>>> _______________________________________________
>>> Asterisk-Users mailing list
>>> Asterisk-Users at lists.digium.com
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>  
>>
>>  
>>
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>
>
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