PROBLEM in making conference room if set to dynamic

Asterisk Asterisk Dialplan
1

Dear team,

Please guide me i am unable to figure it out.

my actual requirement is that:

  1. User A → dials to User B
  2. User A hears ring bell when dialing to User B
  3. When User B lifts the call, both User A and User B will be on the bridge.
  4. And User A clicks on add conference button in mobile and dial to User C ( keeping User B on hold )
  5. Again same when dialing to User C , User A should hear the ringing tone.
  6. And when User C lifts the call then all 3 users i.e A , B and C will be in same call.
  7. Whenever users leaves from bridge, User A ( Admin ) should be notified.
  8. When user A leaves, all other users should leave automatically.

Please find below mentioned code and attached confbridge configuration.

[internal-test]

exten => _X.,1,mixmonitor(/var/spool/asterisk/recording/${STRFTIME(${EPOCH},,%Y-%m-%d)}/${STRFTIME(${EPOCH},,%H-%M)}-${EXTEN}.gsm,b)
same => n,set(CONTACTS=${PJSIP_DIAL_CONTACTS(${EXTEN})})
same => n,gotoif($["${EXTEN:0:2}"="91"]?twilio:normal)
same => n(twilio),execif($["${CONTACTS}"=""]?set(CONTACTS=PJSIP/+${EXTEN}@twilio))
same => n(silence),playback(silence)
same => n,progress()
same => n,ringing()
same => n,set(code=${RAND(100000,999999)})
same => n,dial(${CONTACTS},45,RrG(bridge_test,s,1))
same => n,noop(${DIALSTATUS},${HANGUPCAUSE})
same => n,gotoif($["${HANGUPCAUSE}"="19"]?busy:nxt)
same => n(busy),playback(bussy)
same => n,hangup()
same => n(nxt),gotoif($["${HANGUPCAUSE}"="0"]?no:chn)
same => n(no),playback(noanswer)
same => n,hangup()
same => n(chn),gotoif($["${HANGUPCAUSE}"="21" || "${HANGUPCAUSE}"="3"]?chnx:h,1)
same => n(chnx),playback(channotavailable)
same => n,hangup()
same => n(normal),set(CONTACTS=${PJSIP_DIAL_CONTACTS(${EXTEN})})
same => n,execif($["${CONTACTS}"=""]?set(CONTACTS=PJSIP/+${EXTEN}))
same => n,goto(silence)
exten => h,1,hangup

[bridge_test]

exten => s,1,noop(test bridge ${code})
same =>n,confbridge(< dynamic >,default_bridge,Admins)
same =>n,hangup


i have attached confbridge configuration..
[confbridge.txt|attachment](upload://7eO8NcOQX5Zv4zzUuXWRGEzJnAe.txt) (30.5 KB)

Below mentioned is system configuration

~]# cat /etc/os-release
NAME=“CentOS Stream”
VERSION=“9”
ID=“centos”
ID_LIKE=“rhel fedora”
VERSION_ID=“9”
PLATFORM_ID=“platform:el9”
PRETTY_NAME=“CentOS Stream 9”
ANSI_COLOR=“0;31”
LOGO=“fedora-logo-icon”
CPE_NAME=“cpe:/o:centos:centos:9”
HOME_URL=“https://centos.org/”
BUG_REPORT_URL=“https://issues.redhat.com/”
REDHAT_SUPPORT_PRODUCT=“Red Hat Enterprise Linux 9”
REDHAT_SUPPORT_PRODUCT_VERSION=“CentOS Stream”

My asterisk version is as follows

~]# asterisk -r
Asterisk 20.12.0, Copyright (C) 1999 - 2025, Sangoma Technologies Corporation and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 20.12.0 currently running




please help me. 

thanks and regards
2

This answers the call. Progress is meaningless after than. I’m not sure about Ringing.

3

Dear team,

thank you so much for your reply, i removed progress from my dialplan code.

ringing is required. without this, iam not able to hear ring of callee

please help me on how to proceed with next step

thanks and regards,