@kwriley87 wrote:
Hello I am running FreePBX 12.0.76.4 on Asterisk 11.15.0
I am running into problems installing and using the G729 codec, here are the steps I have taken..
I have checked and confirmed using uname -m that my system is 32 bit.I am downloading this package from the asterisk hosting website: codec_g729-ast110-gcc4-glibc-pentium4.so
cd /usr/lib64/asterisk/modules
wget http://asterisk.hosting.lv/bin/codec_g729-ast110-gcc4-glibc-pentium4.so
mv codec_g729-ast110-gcc4-glibc-pentium4.so codec_g729.so
asterisk -rvvvvvvvvv
module load codec_g729.so
I get an error saying [2016-08-02 14:45:07] WARNING[11670]: loader.c:908 load_resource: Module 'codec_g729.so' already exists.
service asterisk restart
asterisk -rv
core show translation
pbx*CLI> core show translationTranslation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) gsm ulaw alaw g726 adpcm slin lpc10 g729 ilbc g726aal2 g722 slin16 testlaw gsm - 15000 15000 15000 15000 9000 15000 15000 15000 15000 17250 26250 15000 ulaw 15000 - 9150 15000 15000 9000 15000 15000 15000 15000 17250 26250 15000 alaw 15000 9150 - 15000 15000 9000 15000 15000 15000 15000 17250 26250 15000 g726 15000 15000 15000 - 15000 9000 15000 15000 15000 15000 17250 26250 15000 adpcm 15000 15000 15000 15000 - 9000 15000 15000 15000 15000 17250 26250 15000 slin 6000 6000 6000 6000 6000 - 6000 6000 6000 6000 8250 17250 6000 lpc10 15000 15000 15000 15000 15000 9000 - 15000 15000 15000 17250 26250 15000 g729 15000 15000 15000 15000 15000 9000 15000 - 15000 15000 17250 26250 15000 ilbc 15000 15000 15000 15000 15000 9000 15000 15000 - 15000 17250 26250 15000
g726aal2 15000 15000 15000 15000 15000 9000 15000 15000 15000 - 17250 26250 15000
g722 15600 15600 15600 15600 15600 9600 15600 15600 15600 15600 - 9000 15600
slin16 21600 21600 21600 21600 21600 15600 21600 21600 21600 21600 6000 - 21600
testlaw 15000 15000 15000 15000 15000 9000 15000 15000 15000 15000 17250 26250 -
pbx*CLI>Judging by this, G729 appears to be installed correctly.
Back into FreePBX I goto Settings > Asterisk SIP Settings and ensure that G729 is checked.
I go to my primary outbound trunk and confirm that i have allow=all set
I go to my test extension and specifically deny all codecs but allow g729I place a test call and it fails.. excerpt from logs:
-- Executing [s@macro-dialout-trunk:23] NoOp("SIP/800-0000004d", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 18") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/800-0000004d", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/800-0000004d", "RC=18") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/800-0000004d", "18,1") in new stack
-- Executing [18@macro-dialout-trunk:1] Goto("SIP/800-0000004d", "s-NOANSWER,1") in new stack
-- Executing [s-NOANSWER@macro-dialout-trunk:1] NoOp("SIP/800-0000004d", "Dial failed due to trunk reporting NOANSWER - giving up") in new stack
-- Executing [s-NOANSWER@macro-dialout-trunk:2] Progress("SIP/800-0000004d", "") in new stack
-- Executing [s-NOANSWER@macro-dialout-trunk:3] Playback("SIP/800-0000004d", "number-not-answering,noanswer") in new stack
-- Playing 'number-not-answering.gsm' (language 'en')While the call is being placed I can goto Asterisk Info > Channels and see the open G729 channel while I attempt to place the call:
64.136.174.30 12146861983 562001b370a41f0 (g729) No Init: INVITE VOIP_INNOVCan anyone help me figure out what step I am missing here?
Thank you!
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