[Asterisk-Users] Are codec "capabilities bitmasks" different in IAX and SIP?

[Brian Capouch]

I didn't know how else to caption this.

I'm trying to play around with codec pass-through.  I have two SIP 
phones, both with g729, behind two Asterisk servers.

I set all the configs, SIP and IAX, to "disallow=all; allow=g729" on 
both servers.

But the originating server won't even try to call the destination server:

     -- Executing Dial("SIP/btel-c7d7", "IAX2/bris/10101") in new stack
Mar  5 02:55:32 WARNING[2786]: channel.c:1942 ast_request: No translator 
path exists for channel type IAX2 (native 63508) to 256
Mar  5 02:55:32 NOTICE[2786]: app_dial.c:936 dial_exec_full: Unable to 
create channel of type 'IAX2' (cause 0)
   == Everyone is busy/congested at this time (1:0/0/1)
     -- Executing Hangup("SIP/btel-c7d7", "") in new stack
   == Spawn extension (home, 55, 2) exited non-zero on 'SIP/btel-c7d7'

When I show the peer entries on both servers, I see these same values 
for the "codec" strings on either end, but they are *different* for the 
IAX peer than the SIP, e.g. here's a snippet from "show peer:"

iax2 show peer bris
   * Name       : bris
   Secret       : <Set>
<other stuff omitted>
   Codecs       : 0xf900 (g729)
   Codec Order  : (g729)

sip show peer btel
   * Name       : btel
   Secret       : <Set>
<ditto>
   Codecs       : 0x100 (g729)
   Codec Order  : (g729)

**********************

I'm running CVS-HEAD from yesterday.

I get the same result in reverse if I start the call on the other side.

I have run the Wiki and list archives route; followed the advice there 
to a tee (add some lines to the general context in sip.con) but nothing 
seems to yield anything different than the result shown above.

Thanks.

B.