Firstly, I'm running Asterisk 1.4.4 on Solaris 10.
I have several different internal SIP phones all sharing a single IAX2
VoIP channel.
PHONES |- SIP/uLAW --| ASTERISK
|-- IAX2/g729 |VoIP/ISP
The g729 codec has been registered successfully and appears to be
detected by Asterisk
(NOTE: I have changed what I thought might have been sensitive data)
NOTICE[3181]: codec_g729a.c:411 load_module: G.729 transcoding module
version 33, Copyright (C) 1999-2007 Digium, Inc.
NOTICE[3181]: codec_g729a.c:415 load_module: This module is supplied
under a commercial license granted by Digium, Inc.
NOTICE[3181]: codec_g729a.c:416 load_module: Please see the full license
text supplied by the accompanying
NOTICE[3181]: codec_g729a.c:417 load_module: register utility, or ask
for a copy from Digium.
NOTICE[3181]: codec_g729a.c:419 load_module: This product includes
software developed by the OpenSSL Project
NOTICE[3181]: codec_g729a.c:420 load_module: for use in the OpenSSL
Toolkit. (http://www.openssl.org/)
NOTICE[3181]: codec_g729a.c:421 load_module: Copyright (C) 1998-2006 The
OpenSSL Project
== G.729 Host-ID:
XX:XX:XX:XX:XX:XX:XX:XX:XX:XX:XX:XX:XX:XX:XX:XX:XX:XX:XX:XX
== Found license 'G729-123456' providing 1 channels
== Found total of 1 G.729 licenses
== Registered translator 'g729tolin' from format g729 to slin, cost 7
== Registered translator 'lintog729' from format slin to g729, cost 25
codec_g729a.so = (Annex A/B (floating point) G.729 Coder/Decoder
(optimized for i386))
When I make a telephone call the phone rings but as soon as it is
answered it gets dropped as shown below.
Call accepted by 111.222.333.444 (format g729)
Format for call is g729
IAX2/ISP-2 is making progress passing it to SIP/111-12345678
IAX2/ISP-2 answered SIP/111-12345678
WARNING[3061]: channel.c:3222 ast_channel_make_compatible: No path to
translate from SIP/111-12345678(4) to IAX2/ISP-2(256)
WARNING[3061]: app_dial.c:1628 dial_exec_full: Had to drop call because
I couldn't make SIP/111-12345678 compatible with IAX2/ISP-2
Hungup 'IAX2/ISP-2'
Spawn extension (default, 5551234, 1) exited non-zero on 'SIP/111-12345678'
I can use an alternate codec (eg gsm) for the IAX2 connection and it
works fine. Also, incoming calls when the g729 codec is in use work
okay. It appears to just be outgoing calls that fail.
Here is a copy of my iax.conf
[general]
bandwidth=low ; I have tried High here as well
jitterbuffer=yes
tos=ef
register = username:[EMAIL PROTECTED]
[ISP]
disallow=all
allow=g729
type=friend
username=username
secret=password
host=iax.isp.au
auth=md5
context=incoming
qualify=3000
And my sip.conf
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
[111]
tos=reliability
type=friend
username=111
host=dynamic
context=default
reinvite=no
canreinvite=no
secret=password
nat=yes
qualify=yes
rtpkeepalive=30
rtptimeout=90
rtpholdtimeout=120
What am I doing wrong?
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