[Asterisk-Users] Zap to SIP transfer problem

[Barton Hodges]

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? 
>> 
>> 
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users