Re: [Asterisk-Users] Zap to SIP transfer problem

2004-03-05 Thread Eric Wieling
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

RE: [Asterisk-Users] Zap to SIP transfer problem

2004-03-05 Thread 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

RE: [Asterisk-Users] Zap to SIP transfer problem

2004-03-05 Thread 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

RE: [Asterisk-Users] Zap to SIP transfer problem

2004-03-05 Thread 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