Hello
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
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.
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”
thanks…
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
FS_INCLUDES=/usr/include/freeswitch
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.
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?
Hi I being trying to accomplish the installation of the open source G729 codec but is asking for the these folder locations “FS_INCLUDES, FS_MODULES” but there is no place for me to find these folder and there is not a How To guide for teh installation of this codec in ASTPP server, can you please help me and let me know the steps to einstall the open source G729 codec… Thanks
You need to add mod_bcg729.so under /usr/lib/freeswitch/mod/ then load mod_bcg729 from fs_cli and also have it in sip profile as well to offer while the call by freeswitch
thanks for the qucik reply, but the file “mod_bcg729.so” is not present in the codec package and whe I try to compile the codec I got stuck in this error "cc -fPIC -O3 -fomit-frame-pointer -fno-exceptions -Wall -std=c99 -pedantic -I/usr/include -Ibcg729/include -I/usr/include/freeswitch -c mod_bcg729.c
mod_bcg729.c:30:10: fatal error: switch.h: No such file or directory
30 | #include “switch.h” " and have not found a solution to it yet, thanks a lot for your help.
You can directly download it from some 3rd party site, It should be available on wild wide web! And put it into mod directory and you should be able to load it directly from fs_cli
It’s been awhile since I did it. I thought I found it precompiled somewhere but now that I think about it, I may have compiled it myself. It wasn’t too hard as I recall, assuming you have compiled source code before. I am pretty sure I found instructions somewhere.
Hello Sir
I am just wondering do you have any installation guide to for to install g729 codec in my astpp server? one of my provider not supporting other codecs only g729.
can you please help me?
thanks