Confused About Codecs

I have an Odroid HC2 (odroidxu4; armv7l) and I have my sip registered, but I’m getting this error when I try to call from another phone (not part of asterisk):

[Apr 25 12:57:00] NOTICE[27691][C-00000001]: chan_sip.c:10798 process_sdp: No compatible codecs, not accepting this offer!

I think it’s because I don’t have g729 codec installed maybe? But when I type in ffmpeg -codecs, it shows D.A.L. g729 G.729 in the list. I’m unsure if that’s relevant and whether that means I have g729 installed, because I read other stuff online saying that I have to compile g729 for arm chips. I tried following these instructions, but got stuck on when I type make under Compile g279 codecs, I get the error (eventually): codec_g72x.c:21:30: fatal error: asterisk.h: No such file or directory. So I scrapped that idea.

Here is what I have in sip.conf (minus my username and password). Freephoneline.ca said for “used preferred codec” is G711u/PCMU or G729a. Sorry if my sip.conf looks wrong. I tried to keep the defaults what was originally in sip.conf, plus what this person said they did for freephoneline.ca:

[general]
context=public
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes

register => 15555555555:password@voip.freephoneline.ca/15555555555

context=from-trunk
host=voip.freephoneline.ca
username=15555555555
secret=password
type=friend
insecure=port,invite
dtmfmode=auto
disallow=all
allow=ulaw&g729
fromdomain=freephoneline.ca

secret=[password]
type=user
context=from-trunk
insecure=port,invite
fromdomain=freephoneline.ca

registertimeout=3600
registerexpiry=3600

[basic-options](!)
	dtmfmode=rfc2833
	context=from-office
	type=friend

[natted-phone](!,basic-options)
	directmedia=no
	host=dynamic

[public-phone](!,basic-options)
	directmedia=yes

[my-codecs](!)
	disallow=all
	allow=ilbc
	allow=g729
	allow=gsm
	allow=g723
	allow=ulaw

[ulaw-phone](!)
	disallow=all
	allow=ulaw

core show translation and use one of those codec on that list , that the other party also support

The problem is fixed. It was my sip.conf file that needed fixing. Someone fixed it for me and this is what it looks like now. Thanks:

[general]
context=public
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes

register => 15555555555:password@voip.freephoneline.ca/15555555555

[freephoneline]
context=from-trunk
host=voip.freephoneline.ca
username=15555555555
secret=password
type=friend
insecure=port,invite
dtmfmode=auto
disallow=all
allow=ulaw
fromdomain=freephoneline.ca
secret=[password]
type=user
context=from-trunk
insecure=port,invite
fromdomain=freephoneline.ca

[basic-options](!)
    dtmfmode=rfc2833
    context=from-office
    type=friend

[natted-phone](!,basic-options)
    directmedia=no
    host=dynamic

[public-phone](!,basic-options)
    directmedia=yes

[my-codecs](!)
    disallow=all

you removed g729, simle like that

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