Murray Blakeman
2008-Nov-13 11:26 UTC
[asterisk-users] Problems with Licensed g729a codec from Digium
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:password at iax.isp.au [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?