[Asterisk-Users] Codec Issue on IAX trunk?

[Noah Miller]

Hi All -

Well, after happily existing in a one office environment with asterisk 
for a few months, I've now decided to start adding in our other offices 
with their own * boxes and IAX connections (over VPN).  Unfortunately, 
I'm an idiot and I can't get it to work.  I'm having some kind of 
problem with codecs, I guess, but I don't understand what or why.  When 
trying to use an IAX connection to get to another office, I get:

     -- Executing Dial("SIP/68-4ab6", 
"IAX2/ast33:pass at 192.168.1.130/08 at from-sip") in new stack
Feb 11 14:16:58 WARNING[5653]: channel.c:1898 ast_request: No 
translator path exists for channel type IAX2 (native 0) to 4
Feb 11 14:16:58 NOTICE[5653]: app_dial.c:800 dial_exec: Unable to 
create channel of type 'IAX2' (cause 0)

This is particularly confounding because I have all codecs disabled 
except ulaw (all over, sip devices included).  Is it trying to do 
native bridging?  No lo comprendo.

An "iax2 show peers" seems to show that the IAX connection is made 
between the boxes:

ast33*CLI> iax2 show peers
Name/Username    Host                 Mask             Port      Status
ast551           192.168.1.130   (S)  255.255.255.255  4569 (T)  OK (30 
ms)

ast551*CLI> iax2 show peers
Name/Username    Host                 Mask             Port      Status
ast33            192.168.42.130  (S)  255.255.255.255  4569 (T)  OK (30 
ms)


Here's my info:

ast551:  192.168.1.130
ast33:  192.168.42.130
Version: CVS-HEAD-11/03/04-14:59:37 (both boxes)

IAX.CONF on ast551:
[general]
bindport=4569
notransfer=yes
disallow=all
allow=ulaw

[ast33]
type=friend
auth=md5
secret=pass
context=no-callwaiting
host=192.168.42.130
qualify=yes
trunk=yes
disallow=all
allow=ulaw


IAX.CONF on ast33:
[general]
bindport=4569
disallow=all
allow=ulaw

[ast551]
type=friend
auth=md5
secret=pass
context=no-callwaiting
host=192.168.1.130
qualify=yes
trunk=yes
disallow=all
allow=ulaw


EXTENSIONS.CONF on ast33:
[from-sip]
exten => 68,1,Dial(SIP/68,20)
exten => 68,2,Voicemail(u118)
exten => 68,102,Voicemail(b118)
exten => 68,103,Hangup

exten => 
_[012345]X,1,Dial(IAX2/ast33:pass at 192.168.1.130/${EXTEN}@from-sip)

[no-callwaiting]
include => from-sip
include => outgoing


EXTENSIONS.CONF on ast551:
[from-sip]
exten => 19,1,SetGroup(${EXTEN})
exten => 19,2,CheckGroup(1)
exten => 19,103,Goto(19b,1)
exten => 19,3,Dial(SIP/19,20)
exten => 19,4,Voicemail(u18)
exten => 19,5,Hangup

exten => _6X,1,Dial(IAX2/ast551:pass at 192.168.42.130/${EXTEN}@from-sip)

[no-callwaiting]
include => from-sip
include => outgoing


Thanks for any suggestions!
Noah