[Asterisk-Users] Zap to SIP transfer problem

[Eric Wieling]

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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-- 
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"Unofficial Links" section also see
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