[Asterisk-Users] Re: Codec Issue on IAX trunk?

[Noah Miller]

> >      -- 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.
> >
> > 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/${EXTEN}@from-sip)
> >
> > [no-callwaiting]
> > include => from-sip
> > include => outgoing
>
>
> Looks like you're are getting caught with using "friends" instead of 
> peer
> and user.
>
> Try something like this in iax.conf instead:
> [abc-inc] ; inbound connections from remote site
> type=user
> secret=mysecret
> context=from-site2
> disallow=all
> allow=ulaw ; supports only ulaw
> deny=0.0.0.0/0.0.0.0
> permit=1.2.3.0/255.255.255.0  ; tighten security a little bit
>
> [abc-gw] ; outbound connections to remote site
> type=peer
> secret=mysecret
> username=myusername
> host=1.2.3.4
> disallow=all
> allow=ulaw
>
> Then in your dialplan, use something like this to call the remote site:
> exten => _6X,1,Dial(IAX2/myusername at abc-gw/${EXTEN})
>
> and
>
> [from-site2]
> include => local-extns
>
> Note: I type the majority of the above from memory, so there are likely
> some syntax errors in it. But you should get the picture.

Hi Rich -

I'm amazed you can type all that from memory!  Thanks for taking the 
time to do so.


>  Also, a couple
> of people on the list indicated the friends/user/peer code is now
> broken in cvs-head (as of the last couple of days), so if you're
> running current head, that too could be an issue.

My CVS HEAD version is pretty old (11/04), so hopefully that's not the 
issue.

I tried switching from type=friend over to type=user and type=peer.  
The results seem to be the same.  I still get that error:

-- Executing Dial("SIP/69-69ca", "IAX2/ast33-out/08 at no-callwaiting") in 
new stack
Feb 11 16:45:48 WARNING[5828]: channel.c:1898 ast_request: No 
translator path exists for channel type IAX2 (native 0) to 4
Feb 11 16:45:48 NOTICE[5828]: app_dial.c:800 dial_exec: Unable to 
create channel of type 'IAX2' (cause 0)

The part that worries me is the (native 0).  Why is it saying that IAX2 
is native 0?  With all the disallow=all and allow=ulaw, shouldn't it 
also be type 4?

Thanks!
Noah