Issues with asterisk inside docker

Asterisk Asterisk Support
1

Hi All,
for days I’m trying to setup asterisk inside docker-compose. I was able to create environment connect it to trunk but no audio on inbound/outbound calls. Between local peers I have audio but call drops after some time with similar warning for retransmission. I understand that problem is with configuring asterisk to work with NAT settings but I don’t know what else to do.

I don’t have much experience with setting up asterisk, so I really need help with this.

Environment is hosted on AWS instance so trunk and all peers are outside the local network.

Here is my setup:

docker-compose.yml

version: "3.7"
services:
  asterisk:
    build: ./asterisk/
    restart: always
    # environment: 
    #   ASTERISK_UID: 501
    #   ASTERISK_GID: 20
      # ASTERISK_UID: "id -u"
      # ASTERISK_GID: "id -g"
    ports:
      - 5060:5060/udp
      - 5060:5060/tcp
      - 10000-10050:10000-10050/udp
    # network_mode: host
    # privileged: true
    depends_on: 
      - postgres
    entrypoint: /wait-for-it.sh postgres:5432 --
    command: /docker-entrypoint.sh
    volumes: 
      - ./asterisk/volumes/sounds:/var/lib/asterisk/sounds
      - ./asterisk/volumes/keys:/var/lib/asterisk/keys
      - ./asterisk/volumes/phoneprov:/var/lib/asterisk/phoneprov 
      - ./asterisk/volumes/spool:/var/spool/asterisk
      - ./asterisk/volumes/logs:/var/log/asterisk
      - ./asterisk/config/sip.conf:/etc/asterisk/sip.conf
      - ./asterisk/config/queues.conf:/etc/asterisk/queues.conf
      - ./asterisk/config/extensions.conf:/etc/asterisk/extensions.conf
      - ./asterisk/config/cdr.conf:/etc/asterisk/cdr.conf
      - ./asterisk/config/cdr_pgsql.conf:/etc/asterisk/cdr_pgsql.conf
      - ./asterisk/config/rtp.conf:/etc/asterisk/rtp.conf
      - ./asterisk/config/pjsip.conf:/etc/asterisk/pjsip.conf

  postgres:
    build: ./postgres/
    restart: always
    ports:
      - "5432:5432"
    environment:
      POSTGRES_PASSWORD: 123123
      POSTGRES_DB: asterisk
    volumes:
      - postgres-volume:/var/lib/postgresql/data


volumes:
  postgres-volume:
    name: asterisk-postgres-volume

sip.conf

[general]
context=public                  ; Default context for incoming calls. Defaults to 'default'
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
udpbindaddr=0.0.0.0             ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
tcpenable=no                    ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0             ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
transport=transport-udp-nat                   ; Set the default transports.  The order determines the primary default transport.
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
qualify=yes
direct_media=no
rtp_symmetric=yes
force_rport=yes

nat=force_rport,comedia
;localnet=10.0.0.0/255.0.0.0
;localnet=192.168.0.0/255.255.0.0
;externip=18.214.49.215
;107.22.170.158

;register => ******:*****XGPxCE@us-east-va.sip.flowroute.com
;allow=ulaw

[authentication]
[basic-options](!)                ; a template
    dtmfmode=rfc2833
    context=from-office
    type=friend
[natted-phone](!,basic-options)   ; another template inheriting basic-options
    directmedia=no
    host=dynamic
[public-phone](!,basic-options)   ; another template inheriting basic-options
    directmedia=yes
[my-codecs](!)                    ; a template for my preferred codecs
    disallow=all
    allow=ilbc
    allow=g729
    allow=gsm
    allow=g723
    allow=ulaw
[ulaw-phone](!)                   ; and another one for ulaw-only
    disallow=all
    allow=ulaw

[flowroute]
    type=friend
    secret=*********
    username=****8709
    host=us-east-va.sip.flowroute.com
    dtmfmode=rfc2833
    context=incoming
    canreinvite=no
    allow=ulaw
    insecure=port,invite
    fromdomain=us-east-va.sip.flowroute.com

[agent1]
    type=friend
    context=phones
    allow=ulaw,alaw
    secret=123123
    host=dynamic
    ;nat=yes

[agent2]
    type=friend
    context=phones
    allow=ulaw,alaw
    secret=123123
    host=dynamic
    ;nat=yes

[agent3]
    type=friend
    context=phones
    allow=ulaw,alaw
    secret=123123
    host=dynamic
    ;nat=yes

pjsip.conf

[transport-udp-nat]

type=transport

protocol=udp

;bind=0.0.0.0

localnet=10.0.0.0/255.0.0.0
localnet=192.168.0.0/255.255.0.0

;external_media_address=127.0.0.1
;external_signaling_address=127.0.0.1

;type=transport
;protocol=udp
;bind=0.0.0.0
;local_net=192.168.65.0/24
;external_media_address=host.docker.internal
;external_signaling_address=host.docker.internal

external_media_address=18.214.49.215
external_signaling_address=18.214.49.215

extensions.conf

[general]

[globals]

AREACODE= 
;Enter a local 3-digit area code here

TECHPREFIX=****8709* 
;Populate only if IP-based authentication is desired, you’ll need to
;whitelist your IP as well. You must use an eight-digit format, followed by an
;asterix (*)—12345678*.

[outgoing]
exten => _1NXXNXXXXXX,1,Dial(SIP/${TECHPREFIX}${EXTEN}@flowroute)
exten => _NXXNXXXXXX,1,Dial(SIP/${TECHPREFIX}1${EXTEN}@flowroute)
exten => _NXXXXXX,1,Dial(SIP/${TECHPREFIX}1${AREACODE}${EXTEN}@flowroute)
exten => _011.,1,Dial(SIP/${TECHPREFIX}${EXTEN:3}@flowroute)

[incoming]
exten => 12013714710,1,Goto(phones,103,1)

[phones]

exten => 101,1,NoOp(First Line)
same => n,Dial(Sip/agent1)
same => n,Hangup

exten => 102,1,NoOp(First Line)
same => n,Dial(Sip/agent3)
same => n,Hangup

exten => 103,1,NoOp(First Line)
same => n,Dial(Sip/agent3)
same => n,Hangup

exten => _1NXXNXXXXXX,1,Goto(outgoing,${EXTEN},1)

Log:

asterisk_1  | [Oct  6 07:42:13] NOTICE[66]: chan_sip.c:24708 handle_response_peerpoke: Peer 'flowroute' is now Reachable. (1ms / 2000ms)
asterisk_1  | [Oct  6 07:42:16]     -- Registered SIP 'agent1' at 192.168.224.1:60099
asterisk_1  | [Oct  6 07:42:16]     -- Registered SIP 'agent3' at 192.168.224.1:60099
asterisk_1  | [Oct  6 07:42:17] NOTICE[66]: chan_sip.c:24708 handle_response_peerpoke: Peer 'agent1' is now Reachable. (130ms / 2000ms)
asterisk_1  | [Oct  6 07:42:17] NOTICE[66]: chan_sip.c:24708 handle_response_peerpoke: Peer 'agent3' is now Reachable. (131ms / 2000ms)
asterisk_1  | [Oct  6 07:42:46]   == Using SIP RTP CoS mark 5
asterisk_1  | [Oct  6 07:42:46]     -- Executing [15854570651@phones:1] Goto("SIP/agent3-00000000", "outgoing,15854570651,1") in new stack
asterisk_1  | [Oct  6 07:42:46]     -- Goto (outgoing,15854570651,1)
asterisk_1  | [Oct  6 07:42:46]     -- Executing [15854570651@outgoing:1] Dial("SIP/agent3-00000000", "SIP/83758709*15854570651@flowroute") in new stack
asterisk_1  | [Oct  6 07:42:46]   == Using SIP RTP CoS mark 5
asterisk_1  | [Oct  6 07:42:46]     -- Called SIP/****8709*15854570651@flowroute
asterisk_1  | [Oct  6 07:42:48]     -- SIP/flowroute-00000001 is ringing
asterisk_1  | [Oct  6 07:42:48]     -- SIP/flowroute-00000001 is making progress passing it to SIP/agent3-00000000
asterisk_1  | [Oct  6 07:43:12]     -- SIP/flowroute-00000001 answered SIP/agent3-00000000
asterisk_1  | [Oct  6 07:43:12]     -- Channel SIP/flowroute-00000001 joined 'simple_bridge' basic-bridge <18b42844-82cf-476f-bea7-01f126211907>
asterisk_1  | [Oct  6 07:43:12]     -- Channel SIP/agent3-00000000 joined 'simple_bridge' basic-bridge <18b42844-82cf-476f-bea7-01f126211907>
asterisk_1  | [Oct  6 07:43:20] WARNING[66]: chan_sip.c:4072 retrans_pkt: Retransmission timeout reached on transmission nVSrySim9xRuDC-fSdbn8nsu83Zn2U3A for seqno 11771 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
asterisk_1  | Packet timed out after 8385ms with no response
asterisk_1  | [Oct  6 07:43:20] WARNING[66]: chan_sip.c:4096 retrans_pkt: Hanging up call nVSrySim9xRuDC-fSdbn8nsu83Zn2U3A - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
asterisk_1  | [Oct  6 07:43:20]     -- Channel SIP/agent3-00000000 left 'native_rtp' basic-bridge <18b42844-82cf-476f-bea7-01f126211907>
asterisk_1  | [Oct  6 07:43:20]   == Spawn extension (outgoing, 15854570651, 1) exited non-zero on 'SIP/agent3-00000000'
asterisk_1  | [Oct  6 07:43:20]     -- Channel SIP/flowroute-00000001 left 'native_rtp' basic-bridge <18b42844-82cf-476f-bea7-01f126211907>




asterisk_1  | [Oct  6 07:43:52]   == Using SIP RTP CoS mark 5
asterisk_1  | [Oct  6 07:43:52]     -- Executing [103@phones:1] NoOp("SIP/agent1-00000002", "First Line") in new stack
asterisk_1  | [Oct  6 07:43:52]     -- Executing [103@phones:2] Dial("SIP/agent1-00000002", "Sip/agent3") in new stack
asterisk_1  | [Oct  6 07:43:52]   == Using SIP RTP CoS mark 5
asterisk_1  | [Oct  6 07:43:52]     -- Called Sip/agent3
asterisk_1  | [Oct  6 07:43:52]     -- SIP/agent3-00000003 is ringing
asterisk_1  | [Oct  6 07:43:52]     -- SIP/agent3-00000003 is ringing
asterisk_1  | [Oct  6 07:43:54]     -- SIP/agent3-00000003 answered SIP/agent1-00000002
asterisk_1  | [Oct  6 07:43:54]     -- Channel SIP/agent3-00000003 joined 'simple_bridge' basic-bridge <f3e76222-d877-4be6-9afe-c979c3b93b86>
asterisk_1  | [Oct  6 07:43:54]     -- Channel SIP/agent1-00000002 joined 'simple_bridge' basic-bridge <f3e76222-d877-4be6-9afe-c979c3b93b86>
asterisk_1  | [Oct  6 07:44:03] WARNING[66]: chan_sip.c:4072 retrans_pkt: Retransmission timeout reached on transmission BQ.5lH4lAOhZh0jdbRxXcwXLsmVg8Syh for seqno 10611 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
asterisk_1  | Packet timed out after 8320ms with no response
asterisk_1  | [Oct  6 07:44:03] WARNING[66]: chan_sip.c:4096 retrans_pkt: Hanging up call BQ.5lH4lAOhZh0jdbRxXcwXLsmVg8Syh - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
asterisk_1  | [Oct  6 07:44:03]     -- Channel SIP/agent1-00000002 left 'native_rtp' basic-bridge <f3e76222-d877-4be6-9afe-c979c3b93b86>
asterisk_1  | [Oct  6 07:44:03]     -- Channel SIP/agent3-00000003 left 'native_rtp' basic-bridge <f3e76222-d877-4be6-9afe-c979c3b93b86>
asterisk_1  | [Oct  6 07:44:03]   == Spawn extension (phones, 103, 2) exited non-zero on 'SIP/agent1-00000002'

Thank you all!!!

2

to me
i think you should use VM Or Physical
you can check rtp.conf start-end config RTP port and show < asterisk -rx “rtp show setting” > or open debug mode rtp use command < rtp set debug on > look packge RTP send and from IP target
Or you check docker mapping network use command < docker ps > check

3

Like @kokojon14192 said, check rtp.conf. You’re only opening up a small number of ports on the docker side, while asterisk will choose from the range defined in rtp.conf.

And asterisk doesn’t start with low or the lowest value, ports are chosen randomly. That’s a 0.5% chance of using an open port.

rtp.conf
[general]
rtpstart=10000
rtpend=20000
docker-compose.yml
    ports:
      - 10000-10050:10000-10050/udp

I would set ports in docker-conpose.yml to match the asterisk side.

1 Like