Displaying 3 results from an estimated 3 matches for "ast_bridge_dtmf_channel_1".
2005 Oct 17
1
Call transfer - atxfer
...*8
[featuremap]
atxfer => *2
blindxfer => #
disconnect => *0
automon => *1
and when I press *2 console says something like this:
Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 42
(*), at 10.2.20.65
Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got
AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1)
Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge:
Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1
Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read
from SIP/rafal-89b1 (1,42)
-- Attempting native bridge of SIP/andrzej...
2003 Dec 05
0
Native bridging with Polycom 600
...k.
The exact spot where the attempt to natively bridge fails is in rtp.c, line
1281 (CVS from October 8, 2003):
f = ast_read(who);
if (!f || ((f->frametype == AST_FRAME_DTMF) &&
(((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))))
A bit of logging shows the frame f is NULL, so Asterisk thinks one side has
hung up, and gives up on the bridging attempt. Of course, the phones are both
up.
Has anyone gotten these phones to bridge correctly, without the RTP stream
traversing Asterisk? Do I need to update my CVS? I'd...
2006 Jan 23
3
canreinvite always =no * no matter what we try :-(
been testing with a rather simple setup.
The mission is to actually get a reinvite to work on the lan.
I am trying with two sipura phones G.711 codec forced on both
both on the lan no nat no fancy options suchs as tT or H
No matter what we do asterisk hangs on to the media path, how
in the world do I get a reinvite to work where the media path
is actually handled by the two phones on the lan?