[Asterisk-Users] SIP to IAX using G.729

[Garry Taylor]

I am testing the following, and have no G729 codecs installed on my asterisk
-

Firefly [G729] -----> asterisk ------> firefly [G729] which works fine,     
    -- Executing Dial("IAX2/3007 at 3007/3", "IAX2/3005|20|t") in new stack
    -- Called 3005
    -- Call accepted by 192.168.2.20 (format G729A)
    -- Format for call is G729A
    -- IAX2/3005/14 is ringing
    -- Registered to '69.73.19.178', who sees us as 220.255.150.39:4569
    -- IAX2/3005/14 answered IAX2/3007 at 3007/3
    -- Attempting native bridge of IAX2/3007 at 3007/3 and IAX2/3005/14

Then I tested -
Firefly [G729] -----> asterisk ------> Cisco 7960G which fails when I answer
the 7960G, (call drops)
    -- Executing Dial("IAX2/3007 at 3007/1", "SIP/3000|20|t") in new stack
    -- Called 3000
Nov 20 15:47:21 WARNING[3457042]: channel.c:2074
ast_channel_make_compatible: No path to translate from SIP/3000-1b68(4) to
IAX2/3007 at 3007/1(256)
    -- SIP/3000-1b68 is ringing
    -- SIP/3000-1b68 answered IAX2/3007 at 3007/1
Nov 20 15:47:25 WARNING[3457042]: channel.c:2074
ast_channel_make_compatible: No path to translate from IAX2/3007 at 3007/1(256)
to SIP/3000-1b68(8)
Nov 20 15:47:25 WARNING[3457042]: app_dial.c:944 dial_exec: Had to drop call
because I couldn't make IAX2/3007 at 3007/1 compatible with SIP/3000-1b68
  == Spawn extension (internal, 3000, 1) exited non-zero on
'IAX2/3007 at 3007/1'
    -- Hungup 'IAX2/3007 at 3007/1'

Does this mean that I would need to install the G729 codecs into asterisk to
make this work? If so, would that mean that asterisk would attempt the call
from itself to the 7960G using the ALAW (or ULAW) codec and not using the
g729 codec?

I did the due diligence and checked with Mr. Google, and did not find any
answer, so please don't forget to flame me if I am wrong!

Regards
Garry Taylor