No audio on both sides

Dear friends, I am in a situation where I don’t know what to do anymore. I am setting up a new Asterisk central 18.10.0, the previous one was an older version, Issabel Asterisk (Version 11.25.3).

The problem I am encountering is with an IP trunk. When I make a call, the call is established, but there is no audio on both sides. The connection is through a VPN to the provider, which is Copaco.

According to the configuration that can be seen, do you think something is missing or needs to be adjusted or changed?
*** I put between “----” so that just the text is better differentiated

---- a portion of rtp set debug on ----
Got RTP packet from 10.10.10.89:4014 (type 00, seq 021932, ts 399040, len 000160)
Sent RTP packet to 10.14.0.142:47380 (type 00, seq 013913, ts 399040, len 000160)
Got RTP packet from 10.10.10.89:4014 (type 00, seq 021933, ts 399200, len 000160)
Sent RTP packet to 10.14.0.142:47380 (type 00, seq 013914, ts 399200, len 000160)
Got RTP packet from 10.10.10.89:4014 (type 00, seq 021934, ts 399360, len 000160)
Sent RTP packet to 10.14.0.142:47380 (type 00, seq 013915, ts 399360, len 000160)
Got RTP packet from 10.10.10.89:4014 (type 00, seq 021935, ts 399520, len 000160)
Sent RTP packet to 10.14.0.142:47380 (type 00, seq 013916, ts 399520, len 000160)
Got RTP packet from 10.10.10.89:4014 (type 00, seq 021936, ts 399680, len 000160)
Sent RTP packet to 10.14.0.142:47380 (type 00, seq 013917, ts 399680, len 000160)
Got RTP packet from 10.10.10.89:4014 (type 00, seq 021937, ts 399840, len 000160)
Sent RTP packet to 10.14.0.142:47380 (type 00, seq 013918, ts 399840, len 000160)
Got RTP packet from 10.10.10.89:4014 (type 00, seq 021938, ts 400000, len 000160)
Sent RTP packet to 10.14.0.142:47380 (type 00, seq 013919, ts 400000, len 000160)
Got RTP packet from 10.10.10.89:4014 (type 00, seq 021939, ts 400160, len 000160)
Sent RTP packet to 10.14.0.142:47380 (type 00, seq 013920, ts 400160, len 000160)
Got RTP packet from 10.10.10.89:4014 (type 00, seq 021940, ts 400320, len 000160)
Sent RTP packet to 10.14.0.142:47380 (type 00, seq 013921, ts 400320, len 000160)
– Channel PJSIP/thinkcomm-100-00000008 left ‘simple_bridge’ basic-bridge <7b3a6fdb-1e3a-4ddb-a4e1-f1ef732f3cf4>
– Channel PJSIP/copaco-cloud-01-00000009 left ‘simple_bridge’ basic-bridge <7b3a6fdb-1e3a-4ddb-a4e1-f1ef732f3cf4>
== Spawn extension (pbx-thinkcomm, 595961862624, 13) exited non-zero on ‘PJSIP/thinkcomm-100-00000008’
== MixMonitor close filestream (mixed)
== End MixMonitor Recording PJSIP/thinkcomm-100-00000008

---- rtp.conf ----
rtpstart=10000
rtpend=20000

---- pjsip_copaco.conf ----

[copaco-cloud-01]
type=endpoint
transport=transport-udp
aors=copaco-cloud-01
disallow=all
allow=ulaw,alaw
direct_media=no
send_rpid=yes
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
timers=no
from_user=+595217290008
from_domain=copaco.com.py
sdp_owner=+595217290008@copaco.com.py
dtmf_mode=inband
context=pbx-thinkcomm-entrada-copaco-ip

[copaco-cloud-01]
type=aor
contact=sip:10.14.0.141:5060

[copaco-cloud-01]
type=identify
endpoint=copaco-cloud-01
match=10.14.0.141,copaco.com.py

---- pjsip.conf ----
[general]
localnet=10.14.0.0/24
localnet=172.16.71.0/24 ; Red local 1
localnet=10.10.10.60/30 ; Red local 4
externrefresh=10 ; Refresca la IP externa cada 10 segundos

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
local_net=172.16.71.0/24 ; Ajusta tu red local aquí (ya estaba)

[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0

#include pjsip_internos.conf
;#include pjsip_trunks.conf
#include pjsip_copaco.conf

---- Log of making the call from the new pbx 18.10 ----
MixMonitor close filestream (mixed)
== End MixMonitor Recording PJSIP/thinkcomm-100-00000004
– Executing [0961862000@pbx-thinkcomm:1] Goto(“PJSIP/thinkcomm-100-00000006”, “595961862000,1”) in new stack
– Goto (pbx-thinkcomm,595961862000,1)
– Executing [595961862000@pbx-thinkcomm:1] NoOp(“PJSIP/thinkcomm-100-00000006”, “Llamada saliente: Celular”) in new stack
– Executing [595961862000@pbx-thinkcomm:2] Set(“PJSIP/thinkcomm-100-00000006”, “usuario=”) in new stack
– Executing [595961862000@pbx-thinkcomm:3] GotoIf(“PJSIP/thinkcomm-100-00000006”, “1?autenticar”) in new stack
– Goto (pbx-thinkcomm,595961862000,6)
– Executing [595961862000@pbx-thinkcomm:6] Authenticate(“PJSIP/thinkcomm-100-00000006”, “/etc/asterisk/xpinset_2,a,4”) in new stack
– <PJSIP/thinkcomm-100-00000006> Playing ‘agent-pass.gsm’ (language ‘es’)
– <PJSIP/thinkcomm-100-00000006> Playing ‘auth-thankyou.gsm’ (language ‘es’)
– Executing [595961862000@pbx-thinkcomm:7] System(“PJSIP/thinkcomm-100-00000006”, “test -f /etc/asterisk/grabaciones_off/thinkcomm-100”) in new stack
– Executing [595961862000@pbx-thinkcomm:8] GotoIf(“PJSIP/thinkcomm-100-00000006”, “0?discado”) in new stack
– Executing [595961862000@pbx-thinkcomm:9] MixMonitor(“PJSIP/thinkcomm-100-00000006”, “1726860894.6.wav”) in new stack
== Begin MixMonitor Recording PJSIP/thinkcomm-100-00000006
– Executing [595961862000@pbx-thinkcomm:10] Set(“PJSIP/thinkcomm-100-00000006”, “CDR(userfield)=”) in new stack
– Executing [595961862000@pbx-thinkcomm:11] Set(“PJSIP/thinkcomm-100-00000006”, “CDR(ihash)=”) in new stack
– Executing [595961862000@pbx-thinkcomm:12] Progress(“PJSIP/thinkcomm-100-00000006”, “”) in new stack
– Executing [595961862000@pbx-thinkcomm:13] Dial(“PJSIP/thinkcomm-100-00000006”, “PJSIP/0961862000@copaco-cloud-01,120,tT”) in new stack
– Called PJSIP/0961862000@copaco-cloud-01
– PJSIP/copaco-cloud-01-00000007 is making progress passing it to PJSIP/thinkcomm-100-00000006
– PJSIP/copaco-cloud-01-00000007 answered PJSIP/thinkcomm-100-00000006
– Channel PJSIP/copaco-cloud-01-00000007 joined ‘simple_bridge’ basic-bridge <67885f05-7c25-4f83-8402-9a49ea0ec803>
– Channel PJSIP/thinkcomm-100-00000006 joined ‘simple_bridge’ basic-bridge <67885f05-7c25-4f83-8402-9a49ea0ec803>
– Channel PJSIP/thinkcomm-100-00000006 left ‘simple_bridge’ basic-bridge <67885f05-7c25-4f83-8402-9a49ea0ec803>

---- This is a configuration of the previous Isabel that Works ----
---- sip_additional.conf ----
type=peer
session-timers=refuse
qualify=yes
proxy=10.14.0.141
nat=yes
insecure=invite
host=10.14.0.141
fromuser=+595217290008
fromdomain=copaco.com.py
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=yes
canredirect=no
allow=ulaw&alaw&g729

---- sip_nat.conf ----
nat=force_rport,comedia
externip=10.14.2.102
localhost=172.16.71.253
localnet=172.16.71.0/255.255.255.0
localnet=192.168.1.0/255.255.255.0
localnet=192.168.2.0/255.255.255.0
localnet=10.10.10.60/255.255.255.252
qualify=yes
externrefresh=10

You have media flowing in and out of Asterisk in both directions

Where did you find a general section documented. I’m not aware of any documentation that says such a section exists, or how it works.

You don’t appear to have set the external signalling and media addresses on your transport.

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