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
