Displaying 2 results from an estimated 2 matches for "_sip_".
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_sip
2006 Jan 28
1
Can't send DTMF transfer code from called SIP phone
I have several hardware and software phones connected to Asterisk 1.2.1
from Debian via SIP or IAX2 and I have defined call transfer codes in
features.conf. Everything works with the only exception:
When I call a _SIP_ _software_ phone (namely Ekiga or Kphone), I can't
transfer the call from the _callee_ via the configured DTMF codes. It
seems Asterisk completely ignores the sent DTMF codes (no "transfer"
message is received and nothing is written on the log output). Transfer
via the software phon...
2003 Jul 20
1
DTMF crashes chan_capi
...e IVR menu with no problems.
Finally I can bridge the CAPI and SIP channels and hear DTMF digits entered
on the PSTN phone with no problems (they are also detected and displayed on
the console).
However when the CAPI and SIP channels are bridged, entering more than a
couple of DTMF digits into the _SIP_ client appears to crash the channel:
neither party gets disconnected, but there is no longer any audio in either
direction and new calls (inbound or outbound) trying to use the CAPI channel
fail. Once locked if I enter "capi info" in the * console it return nothing
and trying to autocompl...