No RTP engine was found. Do you have one loaded? Asterisk-18.10.1

[xenon52 (xenon5212)]

I have been trying to install Asterisk-18.10.1 version on my ubuntu(20.04.4) running inside VM. I was able to maintain connection from GoTrunk SIP endpoint and Zoiper as softphone. Followed GitHub - GoTrunk/asterisk-config at dynamic-ip tutorial mostly. When I try to call from Zoiper, I got the following error:

I tried module show like rtp , module show like res_rtp_asterisk.so, and module load res_rtp_asterisk.so on Asterisk CLI but couldnot see any of such modules. I checked menuselect, but necessary menus were selected as suggested on previous issues. I also tried: Asterisk 16 has a new module loader ⋆ Asterisk

BUT no progess so far. Would appreciate your direction and any help.

[david55]

See also: No RTP engine was found. Do you have one loaded? Asterisk-18.10.1 - Asterisk Support - Asterisk Community

If you look at the error messages from configure, it is fairly obvious that -with… is not being interpreted as an option, but rather as a positional parameter. Also from the error messages, you can see the correct way of prefixing an option name.

[dicko]

Don’t copy and paste. - is not the same as – (- is not the same as --)

[Stewart1 (Stewart)]

What are you trying to accomplish? If you just want a working FreePBX system under VirtualBox, by far the easiest way is to download the Distro, mount the .iso and you’ll have it running in a few minutes. You’ll also gain several useful features and have the ability to run commercial modules.

Download | FreePBX - Let Freedom Ring

OTOH, if you have a specific need to run FreePBX under Ubuntu, e.g. to run a CRM system in the same image, be aware that FreePBX assumes that it has complete control of its environment and it takes considerable expertise to make another ‘big’ project coexist.

Also, when you set up the new system, I strongly recommend using pjsip and avoiding chan_sip.

[BlazeStudios (Tom Ray)]

1. You have either not built Asterisk with the rtp engine. That means you modified it in menuselect.

2. The RTP module isnt loaded. You might not be loading it right.

3. Something might be stopping it from loading because it isnt installed or not loaded.

4. You dont need to add the with-jasson and with-pjproject-bundled any more. It will grab it with just ./configure but also your commands where wrong. It is two dashes as in --with-pjproject-bundled that is the cause of the build errors.

I would reinstall Asterisk properly then try again.

[dicko]

As of Asterisk 18.10

./configure --help|less

should show that pjproject is ‘bundled’ by default, jansson is not , so first make sure all your ducks are in a row.
From the download directory

contrib/scripts/install_prereq install
./configure --with-jansson-bundled

then when you

make menuselect # adding and checking for  what you feel you need
grep rtp menuselect.makedeps

You should see the ‘dependencies’ are basically PJPROJECT and OPENSSL , so if so, then make && make install should then produce :-

ls /usr/lib*/asterisk/modules/*rtp*

would likely list

bridge_native_rtp.so
chan_rtp.so  
res_pjsip_sdp_rtp.so  
res_rtp_asterisk.so  
res_rtp_multicast.so  
res_srtp.so

Does yours so show?

[xenon52 (xenon5212)]

Yes, I followed along your direction. It does show them.

However, also got warning after make && make install.

When I tried to see if modules have been loaded from Asterisk CLI with module show like rtp, No modules are loaded and still throws the same error.

[xenon52 (xenon5212)]

The warning I got after make && make install was

[dicko]

before you make install issue rm /usr/lib/asterisk/modules/*

[xenon52 (xenon5212)]

seems like I missed to select some important option on make menuselect. Which one should I be most careful with on options?

[dicko]

On a clean install with the ultimate intent to implement FreePBX then you only essentially NEED to add app_macro (which is in the ‘deprecated’ section of apps, go-figure which one Sangoma needs to fix first . . .) and codec specific ‘core’ and ‘extra’ sound files, MOH won’t hurt

[xenon52 (xenon5212)]

The problem in my case might be Asterisk 18 uses PJSIP. However I don’t think I installed PJSIP in my system. I installed v-18, however never installed PJSIP as mentioned in step 2. (tried to setup as if it’s as asterisk 11) . That might be mistake, right?

…ringroost.com/blog/installing-asterisk-13-from-source-on-ubuntu-14/
…gotrunk.com/docs/ippbxconfiguration/asterisk/#dynamic-ip-address
…gotrunk.com/docs/ippbxconfiguration/asteriskpjsipv18/

[dicko]

chan_pjsip would be normally selected for installation on a default install of Asterisk 18.

Try not to do any ‘gefingerpocken’ as you install Asterisk, you WILL need app_macro for FreePBX, start with just that and you should get chan_pjsip and chan_sip compiled and installed.

[Stewart1 (Stewart)]

Wow, 13 posts and no resolution. If you just want to make and receive phone calls, use the Distro; you’ll be making calls in less than an hour. If your application is incompatible with the Distro, I recommend that you browse wiki.freepbx.org and select a cookbook recipe from an expert author, such as
https://wiki.freepbx.org/display/FDT/How+to+Install+FreePBX+16+on+Debian+11+with+Asterisk+16
or
https://wiki.freepbx.org/display/FOP/Installing+FreePBX+16+on+Debian+10.9

If none of them meets your needs, please explain what you are trying to do.

[xenon52 (xenon5212)]

I fixed the RTP Engine problem. I have been trying to use the configuration files from(GitHub - GoTrunk/asterisk-config at dynamic-ip) as I am trying to use Go Trunk SIP trunking service. sip show registry works fine.
The error :

My conf files are here: Conf files - Pastebin.com
LOGS***
[Apr 4 08:58:35] WARNING[732] chan_sip.c: !!! use the global ‘nat’ setting and do not set ‘nat’ per peer/user.
[Apr 4 08:58:35] WARNING[732] chan_sip.c: !!! (config category=‘201’ global force_rport=‘No’ peer/user force_rport=‘Yes’)
[Apr 4 08:58:35] WARNING[732] chan_sip.c: !!! PLEASE NOTE: Setting ‘nat’ for a peer/user that differs from the global setting can make
[Apr 4 08:58:35] WARNING[732] chan_sip.c: !!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users
[Apr 4 08:58:35] WARNING[732] chan_sip.c: !!! will be sent to a different port than replies for an existing peer/user. If at all possible,
[Apr 4 08:58:35] WARNING[732] chan_sip.c: !!! use the global ‘nat’ setting and do not set ‘nat’ per peer/user.
[Apr 4 08:58:35] WARNING[732] chan_sip.c: !!! (config category=‘202’ global force_rport=‘No’ peer/user force_rport=‘Yes’)
[Apr 4 08:58:35] NOTICE[1862] chan_sip.c: Peer ‘202’ is now Reachable. (10ms / 2000ms)
[Apr 4 08:58:35] ERROR[732] ari/config.c: No configured users for ARI
[Apr 4 08:58:35] NOTICE[732] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge
[Apr 4 08:58:35] WARNING[732] cel_pgsql.c: CEL pgsql config file missing global section.
[Apr 4 08:58:35] NOTICE[732] cel_radius.c: Cannot load radiusclient-ng configuration file /etc/radiusclient-ng/radiusclient.conf.
[Apr 4 08:58:35] NOTICE[732] cel_tds.c: cel_tds has no global category, nothing to configure.
[Apr 4 08:58:35] WARNING[732] cel_tds.c: cel_tds module had config problems; declining load
[Apr 4 08:58:35] NOTICE[732] cdr_radius.c: Cannot load radiusclient-ng configuration file /etc/radiusclient-ng/radiusclient.conf.
[Apr 4 08:58:35] NOTICE[732] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Apr 4 08:58:35] NOTICE[732] cdr_pgsql.c: cdr_pgsql configuration contains no global section, skipping module load.
[Apr 4 08:58:35] ERROR[732] chan_unistim.c: Unable to load config unistim.conf
[Apr 4 08:58:35] WARNING[732] res_hep_rtcp.c: res_hep is disabled; declining module load
[Apr 4 08:58:35] ERROR[732] pbx_dundi.c: Unable to load config dundi.conf
[Apr 4 08:58:35] WARNING[732] res_hep_pjsip.c: res_hep is disabled; declining module load
[Apr 4 08:58:35] WARNING[732] loader.c: Some non-required modules failed to load.
[Apr 4 08:58:35] ERROR[732] loader.c: chan_skinny declined to load.
[Apr 4 08:58:35] ERROR[732] loader.c: res_pjsip_transport_websocket declined to load.
[Apr 4 08:58:35] ERROR[732] loader.c: cel_sqlite3_custom declined to load.
[Apr 4 08:58:35] ERROR[732] loader.c: cdr_tds declined to load.
[Apr 4 08:58:35] ERROR[732] loader.c: cel_radius declined to load.
[Apr 4 08:58:35] ERROR[732] loader.c: cel_tds declined to load.
[Apr 4 08:58:35] ERROR[732] loader.c: cdr_radius declined to load.
[Apr 4 08:58:35] ERROR[732] loader.c: cdr_sqlite3_custom declined to load.
[Apr 4 08:58:35] ERROR[732] loader.c: cdr_pgsql declined to load.
[Apr 4 08:58:35] ERROR[732] loader.c: chan_unistim declined to load.
[Apr 4 08:58:35] ERROR[732] loader.c: res_hep_rtcp declined to load.
[Apr 4 08:58:35] ERROR[732] loader.c: pbx_dundi declined to load.
[Apr 4 08:58:35] ERROR[732] loader.c: res_hep_pjsip declined to load.
[Apr 4 08:58:35] NOTICE[1862] chan_sip.c: Peer ‘trunk’ is now Reachable. (69ms / 2000ms)
[Apr 4 09:01:15] NOTICE[1862][C-00000001] chan_sip.c: Call from ‘202’ (192.168.0.163:64136) to extension ‘+15617954831’ rejected because extension not found in context ‘from-internal’.
[Apr 4 09:03:36] NOTICE[1862][C-00000002] chan_sip.c: Call from ‘202’ (192.168.0.163:64136) to extension ‘+15617954831’ rejected because extension not found in context ‘from-internal’.

[dicko]

Using unmaintained for 6 years code which only supports chan_sip and asterisk 13 is not recommended

but good luck if you do.

[xenon52 (xenon5212)]

Which one would you recommend as SIP Trunking provider for easy way set up and be cost effective? I am a completely newbie here. Would appreciate it.

[xenon52 (xenon5212)]

I think I am using asterisk 16 now.

[BlazeStudios (Tom Ray)]

But still using chan_sip which is deprecated and unsupported.