[Asterisk-Dev] Bridge failed on channels

[phu at bevertec.com]

Hi, group:

When I use 2 heterogeneous phones to talk, Asterisk log report the
following problem. I know it is not because asterisk, but I really need
your hints and helps. Thank you very much!

^M  == Parsing '/etc/asterisk/rtp.conf': Found
^M  == RTP Allocating from port range 10000 -> 20000
^MAsterisk Ready.
^[]1;Asterisk^G^[]2;Asterisk Console on 'emv.bevertec.com' (pid
23922)^G*CLI> ^M    -- Got SIP response 481 "Call leg/Transaction does not
exist" back from 192.168.221.72
^M
*CLI> ^M    -- Executing Dial("SIP/2001-44fc", "SIP/2000|20") in new stack
^M    -- Called 2000
^M    -- SIP/2000-7e18 is ringing
^M    -- SIP/2000-7e18 answered SIP/2001-44fc
^M    -- Attempting native bridge of SIP/2001-44fc and SIP/2000-7e18
Mar 16 12:09:48 NOTICE[23922]: channel.c:1724 ast_set_read_format: Unable
to find a path from g723 to ulaw
Mar 16 12:09:48 NOTICE[23922]: channel.c:1691 ast_set_write_format: Unable
to find a path from ulaw to g723
Mar 16 12:09:48 WARNING[23922]: chan_sip.c:1829 sip_write: Asked to
transmit frame type 1, while native formats is 4 (read/write = 4/4)
Mar 16 12:09:48 WARNING[23922]: channel.c:2115
ast_channel_make_compatible: No path to translate from SIP/2001-44fc(1) to
SIP/2000-7e18(4)
Mar 16 12:09:48 WARNING[23922]: channel.c:2641 ast_channel_bridge: Can't
make SIP/2001-44fc and SIP/2000-7e18 compatible
Mar 16 12:09:48 WARNING[23922]: res_features.c:366 ast_bridge_call: Bridge
failed on channels SIP/2001-44fc and SIP/2000-7e18
^M  == Spawn extension (from-sip, 2000, 1) exited non-zero on 'SIP/2001-44fc'
^M
*CLI> !^M


Paul