G729 codec install

i have a new voip supplier. and as usual i try to send calls to his server. but every call was dropping. today he said his server only allow g721,g723 or g729 codec. thats why he got hit but call was not going.
i dont think astpp comes with g721,g723 or g729 codec.
can someone guide me through what to do in this version 4.0.1 version to the g721,g723 or g729 working. personally i dont like g729 as its resource consuming. but if i have no other option then what can i do.

and also i dont want to set it direct in the trunk. i want to set it in the global sip profile… so when a call go system will try to send the call via normal codec but if not will go by g721 or g723 or g729

as i dont want every call go by g729

please someone help


Hello @sokalsondha,
Try once with configuring global codec preference from switch -> sip profiles -> default
from there you can edit “outbound-codec-prefs” & “inbound-codec-prefs” as you need.


Hello sir
I know this how to configure.

I did this. Added g721 and g723 in the sip profile.

But it’s doesn’t work.

Then I add g723 in the in trunk and I can see call going.

But no sound

I did restart and everything. But no luck

Can you tell me what can do?

For RTP-related issues, you can check RTP streams from sip traces using sngrep, if RTP streams are there or not, or you can take pcap file using different tools and check the same in Wireshark.
another thing you can check is, If your server having different interfaces, then check the below params configured with the IP which is reachable from your new voip supplier.
switch -> sip profiles -> default -> “ext-rtp-ip” & “ext-sip-ip”


i think i better install the opensource g729 for transcoding.
but i can’t find any guide…
can you please help me? i am using astpp version 4
i found one in github but its says i need to locate this


when we install astpp where is the FS_INCLUDES location? as i can find anything on /usr/include/freeswitch


Hello @sokalsondha,
If you want to configure G729 codec, I prefer to use the licence version of G729 codec from FreeSWITCH,
sharing the reference URL below.

  1. mod_com_g729 - FreeSWITCH - Confluence
  2. https://freeswitch.com/

okay finally i got sorted.
just one thing… i have sets the codec in the global sip profile.
but if i dont sets the codec thing in the Provider TRUNK its not working.
can we not use from the sip profile instead of for specific trunk?

If there is a space to post a HowTo I can show how install Belledonne G729 Codec on ASTPP.

can you please give me the module file in here… so i can upload it to /usr/lib/freeswitch/mod folder and reload it