[Asterisk-Users] Codec Issue on IAX trunk?

[Rich Adamson]

> 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
 

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. 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.