[asterisk-users] Call failed becaus of SIP tanslate

[khalid touati]

Hi Guys,
I have a the following issue when I ma trying to place a call through my
voip provider, I am using an asterisk 1.2.21.1, do you have an idea what
could fix this issue (as you can see when the other party answered, the call
get dropped because of probably sip incompatibility)

Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit
frame type 256, while native formats is 4 (read/write = 4/4)
Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit
frame type 4, while native formats is 256 (read/write = 256/256)
Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit
frame type 256, while native formats is 4 (read/write = 4/4)
Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit
frame type 4, while native formats is 256 (read/write = 256/256)
Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit
frame type 256, while native formats is 4 (read/write = 4/4)
    -- SIP/to-my-voip-11b955c0 answered SIP/8021-52514588
Nov 12 14:31:30 WARNING[21432]: channel.c:2781 ast_channel_make_compatible:
No path to translate from SIP/8021-52514588(4) to
SIP/to-my-voip-11b955c0(256)
Nov 12 14:31:30 WARNING[21432]: app_dial.c:1628 dial_exec_full: Had to drop
call because I couldn't make SIP/8021-52514588 compatible with
SIP/to-my-voip-11b955c0

Thank you for any help!

-- 
Abdullah
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101112/d7d7c0ef/attachment.htm