[Asterisk-Users] Tranferring UniCall lines

[Steve Underwood]

Guillermo Freige wrote:

> Steve:
> Thanks about the explanation. I'm rather new to all of this digital 
> telephony world. I'm a computer networks guy :)
> If I understood well, the transfer limitation isn´t a MFC/R2 one, but 
> a PSTN one?. Can I transfer calls using the PBX

The limitation is in MFC/R2. Not my implementation of MFC/R2, but in the 
design of the protocol itself. It is not really a PSTN limitation. 
Typically you can do this kind of transfer with the PSTN if you use ISDN 
or SS7.

> call control even in R2 if the PBX support it? Flash didn´t work 
> because it isnt a Zap FXO signalled interface, but

Hook flash will not pass through MFC/R2, but DTMF recall should work. 
However, only some PBXs support that.

> even commenting the condition in the code it didn´t work in the 
> UniCall channel. And Transfer, as said, does nothing at all. If some 
> code is missing in app_flash.c (and the PBX support it) I can patch 
> the code, but I need to know if it is supported by UniCall library (or 
> R2).
> Regarding outgoing calls, I'd 2 problems, the timeout after 1 ring or 
> aroung 4 seconds (it still remains), and a problem with the codec. 
> Apparently alaw wasn´t selected (despite some code tracing showing 
> codec 8 (alaw) was used) and audio was corrupted, and MF codes were 
> sometimes misunderstood because that. I´ve solved it forcing the alaw 
> use in the asterisk code, but now I feel dirty :) The new code will 
> solve this problem too?. Incoming calls work fine (they even show 
> "alaw" as the default codec after a call) and if I use a previously 
> used channel with the codec set for an outgoing call, the sound is OK 
> even with the unmodified code.

I'll check the codec selection. It has been working OK in testing. There 
must be some combination of things where it does wrong.

Regards,
Steve