Not hearing anything accept ringing when call accepted

Asterisk Asterisk Support
1

Hello everyone,

I have an issue with asterisk, when creating a call between 2 devices and accepting it I don’t hear any talking (or other mic noises).
I only hear a ring sound being played (on one device).

I tried finding a solution on the internet but no luck yet, would you be able to help me?

Context
For my Asterisk project I use ARI for call handling (with the c# library AsterNET).
My Asterisk server is hosted with Docker (desktop) on my local machine.

My docker compose for Asterisk:

asterisk:
    ports:
      - 5060:5060/udp               # SIP
      - 8088:8088                   # HTTP
      - 10000-10010:10000-10010/udp # RTP (not the full range to prevent lag with docker desktop)
    image: andrius/asterisk
    restart: always
    volumes:
      - ./logs:/var/log/asterisk
      - ./configs_asterisk:/etc/asterisk

The dailplan I use (extensions.conf):

[public]
exten => _.,1,NoOp()
 same =>    n,Stasis(cocos-stasis-app)
 same =>    n,Hangup()

My pjsip.conf settings:

; Basic UDP transport
;
[transport-udp]
type=transport
protocol=udp    ;udp,tcp,tls,ws,wss,flow
bind=0.0.0.0

[1002]
type=endpoint
auth=1002-auth
aors=1002
disallow=all
allow=alaw
allow=ulaw
context=public
callerid=rick <1002>
dtmf_mode=rfc4733
transport=transport-udp

[1002-auth]
type=auth
auth_type=userpass
password=admin
username=1002

[1002]
type=aor
mailboxes=1002@device
max_contacts=1
remove_existing=yes
maximum_expiration=7200
qualify_frequency=60

[1001]
type=endpoint
context=public
disallow=all
allow=alaw
allow=ulaw
transport=transport-udp
auth=1001-auth
aors=1001

[1001-auth]
type=auth
auth_type=userpass
password=admin
username=1001

[1001]
type=aor
max_contacts=1
mailboxes=1001@device
max_contacts=1
remove_existing=yes
maximum_expiration=7200
qualify_frequency=60

My sip.conf settings:

[general]
context=public
disallow=all ; disallow all codec
allow=alaw
allow=ulaw
localnet=192.168.200.0/21
localnet=127.0.0.0/21
nat=yes
udpbindaddr=0.0.0.0 
transport=udp
srvlookup=yes

My rtp.conf setting:

[general]
rtpstart=10000
rtpend=10010
strictrtp=no

Code that is use to create the call:
Where to.value = 1001 and from.value = 1002

// Create a channel for our from and to devices
Channel channelFrom = Application.ARI.Channels.Create("PJSIP/" + from.value, Application.ARI_STASIS_APP, originator: from.value);
Channel channelTo	= Application.ARI.Channels.Create("PJSIP/" + to.value, Application.ARI_STASIS_APP, originator: to.value);

// Create a bridge where the call can take place
Bridge bridge = Application.ARI.Bridges.CreateAsync("mixing"/*"mixing,dtmf_events"*/, name: $"Bridge_{from.value}{to.value}").Result;

// Add the 2 channels to the bridge
Application.ARI.Bridges.AddChannel(bridge.Id, channelFrom.Id);
Application.ARI.Bridges.AddChannel(bridge.Id, channelTo.Id);

// Make the call
Application.ARI.Channels.Dial(channelTo.Id, channelFrom.Id, 15000);
Application.ARI.Channels.Dial(channelFrom.Id, channelTo.Id, 15000);
2

Although I’m not sure it is the real problem here, you seem to have both chan_psjip and chan_sip trying to bind to port 5060/UDP, which will fail for one of them.

The use of Docker with Asterisk is strongly discouraged, but, again, I’m not sure it is the actual problem.

You haven’t provided any logs, in particular “pjsip set logger on” type logs.

What problem are you trying to solve using ARI, as against the much more well established AMI/Originate method.

3

I don’t know what Stasis does internally, but have you tried adding a Progress() to your incoming context?

Generally that’s needed to get audio to flow before supervision.

4

I think the OP is claiming that both sides are out of early media, but Asterisk is behaving as though they were still in it.

5

Hi, thank you for the reply.

Here are some logs. I set the logger to ON and made the same exact call.

8500018d5477*CLI> pjsip set logger on
PJSIP Logging enabled
    -- Added contact 'sip:1002@192.168.200.179:36627;rinstance=6b783aec544646bf' to AOR '1002' with expiration of 600 seconds
  == Endpoint 1002 is now Reachable
    -- Contact 1002/sip:1002@192.168.200.179:36627;rinstance=6b783aec544646bf is now Reachable.  RTT: 97.715 msec
8500018d5477*CLI>

[Dec  9 08:25:01] NOTICE[103]: res_ari.c:781 process_cors_request: Origin header 'ws://localhost:8088' does not match an allowed origin.
 Creating Stasis app 'stasis-app'
  == WebSocket connection from '172.18.0.1:38060' for protocol '' accepted using version '13'
    -- Channel PJSIP/1002-00000000 joined 'simple_bridge' stasis-bridge <7f08f9e3-d85a-4caa-86f2-c929eb68cfb9>
    -- Channel PJSIP/1001-00000001 joined 'simple_bridge' stasis-bridge <7f08f9e3-d85a-4caa-86f2-c929eb68cfb9>
    -- Channel PJSIP/1002-00000000 left 'simple_bridge' stasis-bridge <7f08f9e3-d85a-4caa-86f2-c929eb68cfb9>
    -- Channel PJSIP/1001-00000001 left 'simple_bridge' stasis-bridge <7f08f9e3-d85a-4caa-86f2-c929eb68cfb9>
8500018d5477*CLI>

O and maybe the /var/log/asterisk/message logs for today would be useful:

[Dec  9 08:19:35] Asterisk 18.11.2 built by buildozer @ build-3-16-x86_64 on a x86_64 running Linux on 2022-04-28 07:42:40 UTC
[Dec  9 08:19:35] NOTICE[1] loader.c: 313 modules will be loaded.
[Dec  9 08:19:36] NOTICE[1] cdr.c: CDR simple logging enabled.
[Dec  9 08:19:36] ERROR[37] res_pjsip/config_system.c: There are no local system nameservers configured, resorting to system resolution
[Dec  9 08:19:36] ERROR[37] res_pjsip/config_system.c: There are no local system nameservers configured, resorting to system resolution
[Dec  9 08:19:36] NOTICE[1] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Dec  9 08:19:36] WARNING[1] res_phoneprov.c: Unable to find a valid server address or name.
[Dec  9 08:19:36] ERROR[1] file.c: Error opening directory - /var/lib/asterisk/moh: No such file or directory
[Dec  9 08:19:36] WARNING[1] res_musiconhold.c: No music on hold classes configured, disabling music on hold.
[Dec  9 08:19:37] NOTICE[1] chan_skinny.c: Configuring skinny from skinny.conf
[Dec  9 08:19:37] NOTICE[1] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge
[Dec  9 08:19:37] NOTICE[1] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Dec  9 08:19:37] WARNING[1] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 12 of extensions.conf
[Dec  9 08:19:37] WARNING[1] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 14 of extensions.conf
[Dec  9 08:19:37] WARNING[1] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 15 of extensions.conf
[Dec  9 08:19:37] WARNING[1] pbx.c: Context 'local' tries to include nonexistent context 'iaxtel700'
[Dec  9 08:19:37] WARNING[1] pbx.c: Context 'local' tries to include nonexistent context 'iaxtel700'
[Dec  9 08:19:37] WARNING[1] loader.c: Some non-required modules failed to load.
[Dec  9 08:19:37] WARNING[1] loader.c: The deprecated module 'res_monitor.so' has been loaded and is running, it may be removed in a future version
[Dec  9 08:19:37] WARNING[1] loader.c: The deprecated module 'res_adsi.so' has been loaded and is running, it may be removed in a future version
[Dec  9 08:19:37] WARNING[1] loader.c: The deprecated module 'app_url.so' has been loaded and is running, it may be removed in a future version
[Dec  9 08:19:37] WARNING[1] loader.c: The deprecated module 'chan_oss.so' has been loaded and is running, it may be removed in a future version
[Dec  9 08:19:37] WARNING[1] loader.c: The deprecated module 'app_image.so' has been loaded and is running, it may be removed in a future version
[Dec  9 08:19:37] WARNING[1] loader.c: The deprecated module 'app_adsiprog.so' has been loaded and is running, it may be removed in a future version
[Dec  9 08:19:37] WARNING[1] loader.c: The deprecated module 'app_getcpeid.so' has been loaded and is running, it may be removed in a future version
[Dec  9 08:19:37] WARNING[1] loader.c: The deprecated module 'app_ices.so' has been loaded and is running, it may be removed in a future version
[Dec  9 08:19:37] WARNING[1] loader.c: The deprecated module 'app_nbscat.so' has been loaded and is running, it may be removed in a future version
[Dec  9 08:19:37] ERROR[1] loader.c: Failed to resolve dependencies for res_stir_shaken
[Dec  9 08:19:37] ERROR[1] loader.c: res_stir_shaken declined to load.
[Dec  9 08:19:37] ERROR[1] loader.c: cel_sqlite3_custom declined to load.
[Dec  9 08:19:37] ERROR[1] loader.c: cdr_csv declined to load.
[Dec  9 08:19:37] ERROR[1] loader.c: cdr_sqlite3_custom declined to load.
[Dec  9 08:19:37] ERROR[1] loader.c: Failed to resolve dependencies for res_http_media_cache
[Dec  9 08:19:37] ERROR[1] loader.c: res_http_media_cache declined to load.
[Dec  9 08:19:37] ERROR[1] loader.c: Failed to resolve dependencies for res_pjsip_stir_shaken
[Dec  9 08:19:37] ERROR[1] loader.c: res_pjsip_stir_shaken declined to load.
[Dec  9 08:25:01] NOTICE[103] res_ari.c: Origin header 'ws://localhost:8088' does not match an allowed origin.

And for the reason I choose ARI:

We have a platform where users will be able to create dailplans for their devices.
And for that I am making a program that will communicate with this platform (will send that a call is incoming and will handle the given response action).
I am not sure if AMI is better for this project but from what I could gather ARI is a better option.

6

You have to enable verbose logging for pjsip set logger on to work properly.

7

O sorry, here is the verbose log:

8179a2b8030e*CLI> pjsip set logger on
PJSIP Logging enabled
8179a2b8030e*CLI> pjsip set logger verbose on
PJSIP Logging to verbose has been enabled
    -- Channel PJSIP/1002-00000002 joined 'simple_bridge' stasis-bridge <374f17d3-2541-4f09-a33b-8ae813b54f59>
    -- Channel PJSIP/1001-00000003 joined 'simple_bridge' stasis-bridge <374f17d3-2541-4f09-a33b-8ae813b54f59>
<--- Transmitting SIP request (910 bytes) to UDP:192.168.200.53:5062 --->
INVITE sip:1001@192.168.200.53:5062 SIP/2.0
Via: SIP/2.0/UDP 172.18.0.3:5060;rport;branch=z9hG4bKPjm-O8GrDtifIPd4v-2V5nvwn2C7OwXSJt
From: "rick" <sip:1002@172.18.0.3>;tag=cv7vXsu.9kvDcGwQReQEI4ERBVbrzHFi
To: <sip:1001@192.168.200.53>
Contact: <sip:asterisk@172.18.0.3:5060>
Call-ID: 9mAxvBCQ0QJEdPvXI6mw0Cx.vmMp2tWn
CSeq: 7223 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.11.2
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 1248000924 1248000924 IN IP4 172.18.0.3
s=Asterisk
c=IN IP4 172.18.0.3
t=0 0
m=audio 10000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (981 bytes) to UDP:192.168.200.179:48142 --->
INVITE sip:1002@192.168.200.179:48142;rinstance=6b783aec544646bf SIP/2.0
Via: SIP/2.0/UDP 172.18.0.3:5060;rport;branch=z9hG4bKPjVvSuOGTBSYShefo.oKp.mSFn3ISsbhCe
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=Aw--HjAzw4q66F27Cf9ysipGClnnwnqU
To: <sip:1002@192.168.200.179;rinstance=6b783aec544646bf>
Contact: <sip:asterisk@172.18.0.3:5060>
Call-ID: WoH9-ZsTLlpc1-pnkXMWnZv7aAfxqLsc
CSeq: 29826 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.11.2
Content-Type: application/sdp
Content-Length:   253

v=0
o=- 42177329 42177329 IN IP4 172.18.0.3
s=Asterisk
c=IN IP4 172.18.0.3
t=0 0
m=audio 10004 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/800<--- Received SIP response (493 bytes) from UDP:192.168.200.53:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.18.0.3:5060;rport=54502;branch=z9hG4bKPjm-O8GrDtifIPd4v-2V5nvwn2C7OwXSJt;received=192.168.200.193
From: "rick" <sip:1002@172.18.0.3>;tag=cv7vXsu.9kvDcGwQReQEI4ERBVbrzHFi
To: <sip:1001@192.168.200.53>
Call-ID: 9mAxvBCQ0QJEdPvXI6mw0Cx.vmMp2tWn
CSeq: 7223 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP1450 1.0.8.9
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (576 bytes) from UDP:192.168.200.53:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.18.0.3:5060;rport=54502;branch=z9hG4bKPjm-O8GrDtifIPd4v-2V5nvwn2C7OwXSJt;received=192.168.200.193
From: "rick" <sip:1002@172.18.0.3>;tag=cv7vXsu.9kvDcGwQReQEI4ERBVbrzHFi
To: <sip:1001@192.168.200.53>;tag=1350510593
Call-ID: 9mAxvBCQ0QJEdPvXI6mw0Cx.vmMp2tWn
CSeq: 7223 INVITE
Contact: <sip:1001@192.168.200.53:5062>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1450 1.0.8.9
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (373 bytes) from UDP:192.168.200.179:48142 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.18.0.3:5060;rport=54502;branch=z9hG4bKPjVvSuOGTBSYShefo.oKp.mSFn3ISsbhCe;received=192.168.200.193
To: <sip:1002@192.168.200.179;rinstance=6b783aec544646bf>
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=Aw--HjAzw4q66F27Cf9ysipGClnnwnqU
Call-ID: WoH9-ZsTLlpc1-pnkXMWnZv7aAfxqLsc
CSeq: 29826 INVITE
Content-Length: 0


<--- Received SIP response (459 bytes) from UDP:192.168.200.179:48142 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.18.0.3:5060;rport=54502;branch=z9hG4bKPjVvSuOGTBSYShefo.oKp.mSFn3ISsbhCe;received=192.168.200.193
Contact: <sip:1002@192.168.200.179:48142>
To: <sip:1002@192.168.200.179;rinstance=6b783aec544646bf>;tag=befa987c
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=Aw--HjAzw4q66F27Cf9ysipGClnnwnqU
Call-ID: WoH9-ZsTLlpc1-pnkXMWnZv7aAfxqLsc
CSeq: 29826 INVITE
User-Agent: SessionTalk 6.0
Content-Length: 0


<--- Received SIP response (459 bytes) from UDP:192.168.200.179:48142 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.18.0.3:5060;rport=54502;branch=z9hG4bKPjVvSuOGTBSYShefo.oKp.mSFn3ISsbhCe;received=192.168.200.193
Contact: <sip:1002@192.168.200.179:48142>
To: <sip:1002@192.168.200.179;rinstance=6b783aec544646bf>;tag=befa987c
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=Aw--HjAzw4q66F27Cf9ysipGClnnwnqU
Call-ID: WoH9-ZsTLlpc1-pnkXMWnZv7aAfxqLsc
CSeq: 29826 INVITE
User-Agent: SessionTalk 6.0
Content-Length: 0


<--- Received SIP response (872 bytes) from UDP:192.168.200.53:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.0.3:5060;rport=54502;branch=z9hG4bKPjm-O8GrDtifIPd4v-2V5nvwn2C7OwXSJt;received=192.168.200.193
From: "rick" <sip:1002@172.18.0.3>;tag=cv7vXsu.9kvDcGwQReQEI4ERBVbrzHFi
To: <sip:1001@192.168.200.53>;tag=1350510593
Call-ID: 9mAxvBCQ0QJEdPvXI6mw0Cx.vmMp2tWn
CSeq: 7223 INVITE
Contact: <sip:1001@192.168.200.53:5062>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1450 1.0.8.9
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length:   239

v=0
o=1001 8001 8000 IN IP4 192.168.200.53
s=SIP Call
c=IN IP4 192.168.200.53
t=0 0
m=audio 5008 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<--- Transmitting SIP request (383 bytes) to UDP:192.168.200.53:5062 --->
ACK sip:1001@192.168.200.53:5062 SIP/2.0
Via: SIP/2.0/UDP 172.18.0.3:5060;rport;branch=z9hG4bKPjsmnYw1rZG4nEH.DuQoGH.p-h82VdhMYV
From: "rick" <sip:1002@172.18.0.3>;tag=cv7vXsu.9kvDcGwQReQEI4ERBVbrzHFi
To: <sip:1001@192.168.200.53>;tag=1350510593
Call-ID: 9mAxvBCQ0QJEdPvXI6mw0Cx.vmMp2tWn
CSeq: 7223 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.11.2
Content-Length:  0


<--- Received SIP response (864 bytes) from UDP:192.168.200.179:48142 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.0.3:5060;rport=54502;branch=z9hG4bKPjVvSuOGTBSYShefo.oKp.mSFn3ISsbhCe;received=192.168.200.193
Require: timer
Contact: <sip:1002@192.168.200.179:48142>
To: <sip:1002@192.168.200.179;rinstance=6b783aec544646bf>;tag=befa987c
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=Aw--HjAzw4q66F27Cf9ysipGClnnwnqU
Call-ID: WoH9-ZsTLlpc1-pnkXMWnZv7aAfxqLsc
CSeq: 29826 INVITE
Session-Expires: 1800;refresher=uac
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE
Content-Type: application/sdp
Supported: path, replaces, timer, norefersub
User-Agent: SessionTalk 6.0
Content-Length: 187

v=0
o=- 0 1 IN IP4 192.168.0.250
s=-
c=IN IP4 192.168.200.179
t=0 0
m=audio 4008 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Transmitting SIP request (429 bytes) to UDP:192.168.200.179:48142 --->
ACK sip:1002@192.168.200.179:48142 SIP/2.0
Via: SIP/2.0/UDP 172.18.0.3:5060;rport;branch=z9hG4bKPjmJuuElDJ37g8QN7KsDMf6xVmoFWG6juJ
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=Aw--HjAzw4q66F27Cf9ysipGClnnwnqU
To: <sip:1002@192.168.200.179;rinstance=6b783aec544646bf>;tag=befa987c
Call-ID: WoH9-ZsTLlpc1-pnkXMWnZv7aAfxqLsc
CSeq: 29826 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.11.2
Content-Length:  0


<--- Transmitting SIP request (407 bytes) to UDP:192.168.200.53:5062 --->
OPTIONS sip:1001@192.168.200.53:5062 SIP/2.0
Via: SIP/2.0/UDP 172.18.0.3:5060;rport;branch=z9hG4bKPjR4TfDzVB4PRFcHkLk5LfOR51YxOoeMGe
From: <sip:1001@172.18.0.3>;tag=Ef--dBGq.hLOiVMZ8zFXs7fapR307ImB
To: <sip:1001@192.168.200.53>
Contact: <sip:1001@172.18.0.3:5060>
Call-ID: nTcTQCrDtDRDj-jxi5dqzpHpzhTksWMp
CSeq: 22173 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.11.2
Content-Length:  0


<--- Received SIP response (498 bytes) from UDP:192.168.200.53:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.0.3:5060;rport=54502;branch=z9hG4bKPjR4TfDzVB4PRFcHkLk5LfOR51YxOoeMGe;received=192.168.200.193
From: <sip:1001@172.18.0.3>;tag=Ef--dBGq.hLOiVMZ8zFXs7fapR307ImB
To: <sip:1001@192.168.200.53>;tag=810731972
Call-ID: nTcTQCrDtDRDj-jxi5dqzpHpzhTksWMp
CSeq: 22173 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1450 1.0.8.9
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP request (464 bytes) to UDP:192.168.200.179:48142 --->
OPTIONS sip:1002@192.168.200.179:48142;rinstance=6b783aec544646bf SIP/2.0
Via: SIP/2.0/UDP 172.18.0.3:5060;rport;branch=z9hG4bKPjpnHzl5T7ZtuHhw4KVS-O1A-g4UiMttkP
From: <sip:1002@172.18.0.3>;tag=KBoshKnbdAajT7-7oFf9KqPXy.IiF7cH
To: <sip:1002@192.168.200.179;rinstance=6b783aec544646bf>
Contact: <sip:1002@172.18.0.3:5060>
Call-ID: VLEGLVwiV0vWHEvTUDMLQJmju7TLwFE7
CSeq: 19653 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.11.2
Content-Length:  0


<--- Received SIP response (588 bytes) from UDP:192.168.200.179:48142 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.0.3:5060;rport=54502;branch=z9hG4bKPjpnHzl5T7ZtuHhw4KVS-O1A-g4UiMttkP;received=192.168.200.193
Contact: <sip:192.168.200.179:48142>
To: <sip:1002@192.168.200.179;rinstance=6b783aec544646bf>;tag=9a9b4353
From: <sip:1002@172.18.0.3>;tag=KBoshKnbdAajT7-7oFf9KqPXy.IiF7cH
Call-ID: VLEGLVwiV0vWHEvTUDMLQJmju7TLwFE7
CSeq: 19653 OPTIONS
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE
Supported: path, replaces, timer, norefersub
User-Agent: SessionTalk 6.0
Content-Length: 0


8179a2b8030e*CLI>
8

Update:

I made the same call with an Asterisk server with the exact same configuration on an VM, and it worked.

So it is probably a problem with Asterisk on docker.

9

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