[Asterisk-Users] more about SIP ...

[Dave Alan Caruana]

I added the line "allow G723.1" in my sip.conf general config,
and from a bridge connection which gives silence,
I have progressed to the error message below,
and the call gets rejected.

help!!

Dave

ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant
Expressa
     723 at 216.52.153.207 : Go2Call SIP gateway



    -- Executing Dial("SIP/217.168.168.49:5060", "SIP/723 at 216.52.153.207")
in new stack
    -- Called 723 at 216.52.153.207
WARNING[1240577216]: File channel.c, Line 1711
(ast_channel_make_compatible): No path to translate from
SIP/216.52.153.207-2e12(1) to SIP/217.168.168.49:5060(4)
    -- SIP/216.52.153.207-2e12 answered SIP/217.168.168.49:5060
WARNING[1240577216]: File channel.c, Line 1711
(ast_channel_make_compatible): No path to translate from
SIP/217.168.168.49:5060(4) to SIP/216.52.153.207-2e12(1)
WARNING[1240577216]: File app_dial.c, Line 606 (dial_exec): Had to drop call
because I couldn't make SIP/217.168.168.49:5060 compatible with
SIP/216.52.153.207-2e12
  == Spawn extension (default, 1303, 1) exited non-zero on
'SIP/217.168.168.49:5