Asterisk ERROR[25967] res_pjsip.c: Unable to retrieve PJSIP transport 'transport-udp' since upgrade to 3.1.5-4

Reporting an Issue
1

pbx-a.bellqconnect.nl Carrier Plus 3.1.5-4
Asterisk 18.6.0-1
DAHDI 2.11.1-7

Asterisk ERROR[25967] res_pjsip.c: Unable to retrieve PJSIP transport ‘transport-udp’ since upgrade to 3.1.5-4

Asterisk stopped working correcly, after a reset of asterisk it works again, but I see errors.
In the GUI there is a new Item under “Technology Settings” called “PJSIP Transports”.
In “PJSIP Transports” there are no configs.

I notice that there is an extra file “/etc/asterisk/vitalpbx/pjsip__40-1-profiles.conf”
The file has configs:

;Default PJSIP Profile - Default PJSIP Profile
[p1](!)
media_encryption=no
dtls_setup=actpass
force_rport=yes
ice_support=no
rtp_symmetric=yes
rewrite_contact=yes
use_avpf=no
direct_media=no
disable_direct_media_on_nat=yes
dtls_verify=no
rtcp_mux=no
media_use_received_transport=yes
media_encryption_optimistic=yes
asymmetric_rtp_codec=no
dtls_fingerprint=sha-256
dtls_rekey=0
send_diversion=yes
send_pai=yes
send_rpid=yes
webrtc=no
max_video_streams=5
max_audio_streams=5
transport=transport-udp-3baa003c2d301de89c68

;Default PJSIP Profile - Default PJSIP Profile
[p1-aor](!)
qualify_frequency=30
qualify_timeout=3
remove_existing=yes
support_path=no
default_expiration=3600
maximum_expiration=7200
minimum_expiration=600

;Default WebRTC Profile - Default WebRTC Profile
[p12](!)
media_encryption=dtls
dtls_setup=actpass
force_rport=yes
ice_support=yes
rtp_symmetric=yes
rewrite_contact=no
use_avpf=yes
direct_media=yes
disable_direct_media_on_nat=yes
dtls_verify=fingerprint
rtcp_mux=yes
media_use_received_transport=yes
media_encryption_optimistic=no
asymmetric_rtp_codec=yes
dtls_fingerprint=sha-256
dtls_rekey=0
send_diversion=yes
send_pai=yes
send_rpid=yes
webrtc=yes
max_video_streams=5
max_audio_streams=5
transport=transport-wss-eb7e9c0a1b729f563090

;Default WebRTC Profile - Default WebRTC Profile
[p12-aor](!)
qualify_frequency=30
qualify_timeout=3
remove_existing=yes
support_path=no
default_expiration=3600
maximum_expiration=7200
minimum_expiration=600

;TCP PJSIP Profile - 
[p14](!)
media_encryption=no
dtls_setup=actpass
force_rport=yes
ice_support=no
rtp_symmetric=yes
rewrite_contact=yes
use_avpf=no
direct_media=no
disable_direct_media_on_nat=yes
dtls_verify=no
rtcp_mux=no
media_use_received_transport=yes
media_encryption_optimistic=yes
asymmetric_rtp_codec=no
dtls_fingerprint=sha-256
dtls_rekey=0
send_diversion=yes
send_pai=yes
send_rpid=yes
webrtc=no
max_video_streams=5
max_audio_streams=5
transport=transport-tcp-37eda84c495b784c1dd0

;TCP PJSIP Profile - 
[p14-aor](!)
qualify_frequency=30
qualify_timeout=3
remove_existing=yes
support_path=no
default_expiration=3600
maximum_expiration=7200
minimum_expiration=600

Edit: Changed the text to use preformatted text for easy read - mod

2

The transport “transport-udp” doesn’t exist anymore.

Now, I am not clear about what the error is. Can you please explain in detail what is the error you are having?

3

After setting all trunks of all the tennants to disable en then back to enable the problems seems solved.
The problem was that on incomming calls from trunks this error was created in asterisk and asterisk refused the calls.

4

Can you explain that in more detail please?!

I updated to latest VitalPBX 3.1.6-1 with Asterisk 18.10.0

Created a new PJSIP Devices and got “rejected” when trying to dial out. Internal numbers same thing: REJECTED.
Dialing in was never a problem.
I changed to TCP instead of UDP on the Fanvil X6U and then i was able to dial out.
But then: PJSIP Endpoints in PBX Reports show RED on Contacts because of TCP.

“Default PJSIP Profile” transport is still UDP like it always was.
MicroSIP Softphone is working fine on UDP. This Fanvil X6U even after factory defaults not working fine.

Sound like this could effect many in the future so this should be taken seriously.

5

The transports are now created dynamically. What I meant is that the transport with the name “transport-udp” doesn’t exist. Instead, there’s a UDP transport with a random name.

If you execute the command below, you will see the list of available transports.

asterisk -rx"pjsip show transports"

Result:

Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress....................>
==========================================================================================

Transport:  transport-tcp-37eda84c495b784c1dd0     tcp      0      0  0.0.0.0:5060
Transport:  transport-tls-608c7bb6130ab9815fbe     tls      0      0  0.0.0.0:5061
Transport:  transport-tls-ms-teams-bef5de028ca2dda67de6     tls      0      0  0.0.0.0:5061
Transport:  transport-udp-3baa003c2d301de89c68     udp      0      0  0.0.0.0:5060
Transport:  transport-ws-4a22e66ed2320d0b84e8      ws      0      0  0.0.0.0:5060
Transport:  transport-wss-eb7e9c0a1b729f563090     wss      0      0  0.0.0.0:5060

Objects found: 6

In this case, the UDP transport is “transport-udp-3baa003c2d301de89c68”

1 Like
6

The problem was that the update did not update the asterisk files for the tenants.
After disable and enable the trunks for the tenants new asterisk files where made and the problem was solved. It could have been caused by a copy of a http conf file (my fault) which caused an error after the update (double ports) so the http server did not start automatically after the update.
Maybe the asterisk files for the tenants where not renewed due to the http server was not working?

1 Like
7

Just a quick feedback:
must be an issue with this Fanvil phone not with VitalPBX. Fanvil X6U has been nice but this one phone is driving me nuts. If there are strange things with Fanvil phones even after factory defaults: try a different fanvil phone.