[Asterisk-Users] Zap to SIP transfer problem

[Barton Hodges]

exten => s,10,Dial(${ARG1}/${DIALED},19,Ttm) which translates to
Dial("SIP/210-80f2", "SIP/280|19|Ttm") 

I believe the problem is related to the Grandstream HandyTone-286.  A
caller can transfer, but a callee cannot.  The problem does not exist
with a BT101 (1.0.4.23).  I just tried all of the firmware on their
BETA site, from 1.0.4.35 through 1.0.4.50 and the problem was never
solved.

Can anyone confirm this for me?

I am SO SICK of dealing with HT-286 firmware bugs!

asterisk-users-admin at lists.digium.com wrote:
> What is your ACTUAL Dial line?
> 
> On Fri, 2004-03-05 at 21:19, Barton Hodges wrote:
>> I'm using SIP INFO and ulaw.  It seems that the same thing happens
>> from SIP to SIP as well, regardless of what the DTMF setting is.
The
>> actual problem is that the calling user can transfer, but the
called
>> user cannot.  I just tried the latest CVS snapshot and the v1.0
>> stable branch and they both behave the same way.
>> 
>> asterisk-users-admin at lists.digium.com wrote:
>>> Maybe you are using inband DTMF with a compressed codec. DTMF on a
>>> call with any codec other than ulaw or alaw MUST use OOB DTMF like
>>> RFC2833 or INFO. 
>>> 
>>> On Fri, 2004-03-05 at 20:39, Barton Hodges wrote:
>>>> I'm having a problem with transferring a call that comes in a Zap
>>>> channel and is connected with a SIP channel (on a GS HT-286).
>>>> 
>>>> The call is answered automatically, then the user enters an
>>>> extension. Dial() is called with both T and t flags.  When the
>>>> bridge is made between the channels, the caller on the Zap
channel
>>>> can hit '#' to transfer, but the caller on the SIP channel
cannot.
>>>> No messages whatsoever are displayed on the console when the SIP
>>>> user hits any keys.  What am I missing?
>>>> 
>>>> 
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>> 
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