[asterisk-users] Random "unknown" codec format IAX calls

[Max Ochoa]

    I seem to be having a problem that I have narrowed down to a 
disagreement on codec negotiation or codec setup of some kind in an IAX 
peering arrangement. Here's a non-ASCII art version of the setup:

DID origination provider
  via SIP/gsm
      to
Call routing asterisk server
  via IAX/gsm
      to
Client asterisk server
  via SIP/ulaw
      to
Polycom 501 UA

The problem that occurs sporadically (1/10 times) is the call will 
complete and stay active, but there is no audio. There is a channel open 
all the way to the phone, and the codec (gsm) is shown as the format for 
the call for the SIP channel and the IAX channel from the Call routing 
server to the Client asterisk server. However, the Client asterisk 
server shows the call format as "unknown" when a call is open that has 
no audio. The codec was originally forced to to gsm, then forced to 
ulaw, then set for any (allow=all) with the same results. Here's the 
output from the console on calls with these symptoms. IAX debug output 
looked the same for calls that had audio and those that did not, so I'll 
spare posting that.

Asterisk console (verbose):
=======================================
  -- Accepting AUTHENTICATED call from 10.3.0.1 <http://10.3.0.1>:
       > requested format = gsm,
       > requested prefs = (),
       > actual format = gsm,
       > host prefs = (ulaw|alaw|gsm),
       > priority = mine
    -- Executing Wait("IAX2/customer-8", "0") in new stack
    -- Executing Set("IAX2/customer-8", "_CONTEXTNAME=customer") in new 
stack
    -- Executing Set("IAX2/customer-8", "_VMEXTEN=100") in new stack
    -- Executing Set("IAX2/customer-8", 
"_VOIP_SERVER=customer.voip.domain.net 
<http://brasovan.voip.bestserversllc.net>") in new stack
    -- Executing Set("IAX2/customer-8", "TIMEOUT(digit)=5") in new stack
    -- Digit timeout set to 5
    -- Executing Set("IAX2/customer-8", "TIMEOUT(response)=6") in new stack
    -- Response timeout set to 6
    -- Executing Dial("IAX2/customer-8", "SIP/100-customer&SIP/101
-customer|25|tr") in new stack
    -- Called 100-customer
    -- Called 101-customer
    -- SIP/100-customer-081940e0 is ringing
    -- SIP/101-customer-081a1c70 is ringing
    -- SIP/100-customer-081940e0 answered IAX2/customer-8

customer*CLI> iax2 show channels
Channel               Peer             Username    ID (Lo/Rem)  Seq 
(Tx/Rx)  Lag      Jitter  JitBuf  Format
IAX2/customer-8       10.3.0.1 <http://10.3.0.1>         customer    
00008/00003  00014/00010  00079ms  -0001ms  0000ms  unknow
1 active IAX channel

customer*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     
Last Message   
10.0.0.103 <http://10.0.0.103>       100-custo  54eb074262d  
00102/00000  ulaw  No       Tx: ACK        
1 active SIP channel
===========================

There is a vtun IP tunnel between the Call routing asterisk server and 
the Client asterisk server (the 10.3.0.0/24 subnet.) The 10.0.0.0/24 
subnet is the client's LAN.

Any tips / ideas on what to try next are appreciated.

- Max