Issue: asterisk-pjsip over webrtc

Asterisk Asterisk SIP
1

please check my configurations and logs and help me

[1234567890]
type=endpoint
context=from-internal
disallow=all
allow=ulaw,alaw,g729
auth=auth_1234567890
aors=1234567890
allow=ulaw
direct_media=no
force_rport=yes
rewrite_contact=yes
rtp_symmetric=yes

webrtc=yes
media_encryption=dtls
dtls_verify=fingerprint
dtls_setup=actpass
dtls_rekey=0
ice_support=yes

[1234567890]
type=aor
max_contacts=50
remove_existing=yes
qualify_frequency=30

[auth_1234567890]
type=auth
auth_type=userpass
username=1234567890
password=123

[transport-ws]
type=transport
protocol=ws
bind=0.0.0.0
allow_reload=yes

[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0:8089
cert_file=/etc/letsencrypt/live/OMMS.sytes.net-0001/fullchain.pem
priv_key_file=/etc/letsencrypt/live/OMMS.sytes.net-0001/privkey.pem

external_media_address=OMMS.sytes.net
external_signaling_address=OMMS.sytes.net
allow_reload=yes

[twilio_outbound]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
transport=transport-udp
outbound_auth=twilio_auth
aors=twilio_outbound
send_pai=yes
direct_media=no

dtmf_mode=rfc4733
send_pai=yes
send_rpid=yes
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes

[twilio_outbound]
type=aor
contact=sip:OM.pstn.twilio.com

[twilio_auth]
type=auth
auth_type=userpass

[twilio_auth]
type=auth
auth_type=userpass
username=calling
password=call@1234

[twilio_inbound]
type=endpoint
context=from-twilio
disallow=all
allow=ulaw,alaw
transport=transport-udp
aors=twilio_inbound
auth=twilio_auth
direct_media=no
trust_id_inbound=yes

[twilio_inbound]
type=aor
max_contacts=20

[twilio_auth_inbound]
type=identify
endpoint=twilio_inbound
match=54.172.60.0/30
match=54.244.51.0/30
match=54.171.127.192/30
match=54.65.63.192/30

my extensions.conf file contents are as follows:

[from-internal]

exten => 1234567890,1,Dial(PJSIP/+919978747153@twilio_outbound)

issue is that, call is ringing but when answered the call it disconnects call automatically

below mentioned is my asterisk console log for your reference

*CLI>
– Executing [1234567890@from-internal:1] Dial(“PJSIP/1234567890-00000008”, “PJSIP/+919978747153@twilio_outbound”) in new stack
– Called PJSIP/+919978747153@twilio_outbound
*CLI>
– PJSIP/twilio_outbound-00000009 is making progress passing it to PJSIP/1234567890-00000008
– Call on PJSIP/twilio_outbound-00000009 placed on hold
– Started music on hold, class ‘default’, on channel ‘PJSIP/twilio_outbound-00000009’
– PJSIP/twilio_outbound-00000009 is ringing
– PJSIP/twilio_outbound-00000009 requested media update control 26, passing it to PJSIP/1234567890-00000008
– PJSIP/twilio_outbound-00000009 answered PJSIP/1234567890-00000008
– Channel PJSIP/twilio_outbound-00000009 joined ‘simple_bridge’ basic-bridge
– Channel PJSIP/1234567890-00000008 joined ‘simple_bridge’ basic-bridge
– Stopped music on hold on PJSIP/twilio_outbound-00000009
– Channel PJSIP/1234567890-00000008 left ‘simple_bridge’ basic-bridge
== Spawn extension (from-internal, 1234567890, 1) exited non-zero on ‘PJSIP/1234567890-00000008’
– Channel PJSIP/twilio_outbound-00000009 left ‘simple_bridge’ basic-bridge
*CLI>

please guide me.

thanks and regards

2

You haven’t provided a SIP trace using “pjsip set logger on” to show the SIP signaling, that is needed to see who and what terminates the call.

3

many thanks for your reply please check below logs

PJSIP Logging enabled
<--- Received SIP request (1811 bytes) from WSS:49.49.49.49:40531 --->
INVITE sip:1234567890@OMMS.sytes.net SIP/2.0
Via: SIP/2.0/WSS ffuq9ma9p2hg.invalid;branch=z9hG4bK8340323
Max-Forwards: 70
To: <sip:1234567890@OMMS.sytes.net>
From: <sip:1234567890@OMMS.sytes.net>;tag=ol1746ig2s
Call-ID: kdogmfmrbr2kr08m53ps
CSeq: 8821 INVITE
Contact: <sip:vdfeh0h9@ffuq9ma9p2hg.invalid;transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Type: application/sdp
Content-Length: 1295

v=0
o=- 1960213914208641035 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS aeec8b09-c7fa-4265-8e2d-dc9d85e84b3e
m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:f3wu
a=ice-pwd:6rjcV9URrmAuodtIylbV1WT3
a=ice-options:trickle
a=fingerprint:sha-256 B3:66:AB:AA:E4:4C:10:AE:28:5E:4C:61:DF:9C:45:5E:3C:D0:F2:76:65:CA:08:E4:0A:FC:48:01:33:6D:40:3B
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:aeec8b09-c7fa-4265-8e2d-dc9d85e84b3e 2e1acb68-2cbf-4312-ab12-c1b855ad00ee
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:2757806931 cname:/tVQnrg5hcHK3eMC
a=ssrc:2757806931 msid:aeec8b09-c7fa-4265-8e2d-dc9d85e84b3e 2e1acb68-2cbf-4312-ab12-c1b855ad00ee

<--- Transmitting SIP response (494 bytes) to WSS:49.49.49.49:40531 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS ffuq9ma9p2hg.invalid;rport=40531;received=49.49.49.49;branch=z9hG4bK8340323
Call-ID: kdogmfmrbr2kr08m53ps
From: <sip:1234567890@OMMS.sytes.net>;tag=ol1746ig2s
To: <sip:1234567890@OMMS.sytes.net>;tag=z9hG4bK8340323
CSeq: 8821 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1739865531/56fa6c35e46f8d6ad03af810caf87c64",opaque="0611e37d6db0ee12",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.12.0
Content-Length:  0


<--- Received SIP request (369 bytes) from WSS:49.49.49.49:40531 --->
ACK sip:1234567890@OMMS.sytes.net SIP/2.0
Via: SIP/2.0/WSS ffuq9ma9p2hg.invalid;branch=z9hG4bK5899633
Max-Forwards: 70
To: <sip:1234567890@OMMS.sytes.net>;tag=z9hG4bK8340323
From: <sip:1234567890@OMMS.sytes.net>;tag=ol1746ig2s
Call-ID: kdogmfmrbr2kr08m53ps
CSeq: 8821 ACK
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Length: 0


<--- Received SIP request (2105 bytes) from WSS:49.49.49.49:40531 --->
INVITE sip:1234567890@OMMS.sytes.net SIP/2.0
Via: SIP/2.0/WSS ffuq9ma9p2hg.invalid;branch=z9hG4bK8601973
Max-Forwards: 70
To: <sip:1234567890@OMMS.sytes.net>
From: <sip:1234567890@OMMS.sytes.net>;tag=ol1746ig2s
Call-ID: kdogmfmrbr2kr08m53ps
CSeq: 8822 INVITE
Authorization: Digest algorithm=MD5, username="1234567890", realm="asterisk", nonce="1739865531/56fa6c35e46f8d6ad03af810caf87c64", uri="sip:1234567890@OMMS.sytes.net", response="f0b993cae9ebcf383aa06d290c315ee3", opaque="0611e37d6db0ee12", qop=auth, cnonce="6ru7edoro4io", nc=00000001
Contact: <sip:vdfeh0h9@ffuq9ma9p2hg.invalid;transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Type: application/sdp
Content-Length: 1295

v=0
o=- 1960213914208641035 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS aeec8b09-c7fa-4265-8e2d-dc9d85e84b3e
m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:f3wu
a=ice-pwd:6rjcV9URrmAuodtIylbV1WT3
a=ice-options:trickle
a=fingerprint:sha-256 B3:66:AB:AA:E4:4C:10:AE:28:5E:4C:61:DF:9C:45:5E:3C:D0:F2:76:65:CA:08:E4:0A:FC:48:01:33:6D:40:3B
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:aeec8b09-c7fa-4265-8e2d-dc9d85e84b3e 2e1acb68-2cbf-4312-ab12-c1b855ad00ee
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:2757806931 cname:/tVQnrg5hcHK3eMC
a=ssrc:2757806931 msid:aeec8b09-c7fa-4265-8e2d-dc9d85e84b3e 2e1acb68-2cbf-4312-ab12-c1b855ad00ee

<--- Transmitting SIP response (323 bytes) to WSS:49.49.49.49:40531 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS ffuq9ma9p2hg.invalid;rport=40531;received=49.49.49.49;branch=z9hG4bK8601973
Call-ID: kdogmfmrbr2kr08m53ps
From: <sip:1234567890@OMMS.sytes.net>;tag=ol1746ig2s
To: <sip:1234567890@OMMS.sytes.net>
CSeq: 8822 INVITE
Server: Asterisk PBX 20.12.0
Content-Length:  0


    -- Executing [1234567890@from-internal:1] Dial("PJSIP/1234567890-00000012", "PJSIP/+919978747153@twilio_outbound") in new stack
    -- Called PJSIP/+919978747153@twilio_outbound
<--- Transmitting SIP request (1095 bytes) to UDP:54.172.60.0:5060 --->
INVITE sip:+919978747153@OM.pstn.twilio.com SIP/2.0
Via: SIP/2.0/UDP 94.94.94.94:8631;rport;branch=z9hG4bKPjf9208326-4160-4182-a891-fab981074e9d
From: <sip:1234567890@94.94.94.94>;tag=e5ba1d96-1c5e-4896-94e8-e2c0b406056e
To: <sip:+919978747153@OM.pstn.twilio.com>
Contact: <sip:asterisk@94.94.94.94:8631>
Call-ID: fe7639e5-3494-4ff9-9272-316e9f1269c3
CSeq: 26366 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:1234567890@94.94.94.94>
Remote-Party-ID: <sip:1234567890@94.94.94.94>;party=calling;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 20.12.0
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 1841926769 1841926769 IN IP4 94.94.94.94
s=Asterisk
c=IN IP4 94.94.94.94
t=0 0
m=audio 15320 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (419 bytes) from UDP:54.172.60.0:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 94.94.94.94:8631;rport=8631;branch=z9hG4bKPjf9208326-4160-4182-a891-fab981074e9d;received=94.94.94.94
From: <sip:1234567890@94.94.94.94>;tag=e5ba1d96-1c5e-4896-94e8-e2c0b406056e
To: <sip:+919978747153@OM.pstn.twilio.com>
Call-ID: fe7639e5-3494-4ff9-9272-316e9f1269c3
CSeq: 26366 INVITE
Server: Twilio Gateway
Content-Length: 0


<--- Received SIP response (657 bytes) from UDP:54.172.60.0:5060 --->
SIP/2.0 407 Proxy Authentication required
CSeq: 26366 INVITE
Call-ID: fe7639e5-3494-4ff9-9272-316e9f1269c3
From: <sip:1234567890@94.94.94.94>;tag=e5ba1d96-1c5e-4896-94e8-e2c0b406056e
To: <sip:+919978747153@OM.pstn.twilio.com>;tag=92170301_c3356d0b_4af7de19-9efa-47ab-9f96-81d4543efcd4
Via: SIP/2.0/UDP 94.94.94.94:8631;received=94.94.94.94;rport=8631;branch=z9hG4bKPjf9208326-4160-4182-a891-fab981074e9d
Server: Twilio
Contact: <sip:172.25.44.232:5060>
Proxy-Authenticate: Digest realm="sip.twilio.com",qop="auth",nonce="pHZpzI0OjeXtRh6ZN0BHNiwHXk2v45-1XTFjt0ZVkU29UnDK",opaque="0d975d0cc5e2e4777859904b9029d3aa"
Content-Length: 0


<--- Transmitting SIP request (481 bytes) to UDP:54.172.60.0:5060 --->
ACK sip:+919978747153@OM.pstn.twilio.com SIP/2.0
Via: SIP/2.0/UDP 94.94.94.94:8631;rport;branch=z9hG4bKPjf9208326-4160-4182-a891-fab981074e9d
From: <sip:1234567890@94.94.94.94>;tag=e5ba1d96-1c5e-4896-94e8-e2c0b406056e
To: <sip:+919978747153@OM.pstn.twilio.com>;tag=92170301_c3356d0b_4af7de19-9efa-47ab-9f96-81d4543efcd4
Call-ID: fe7639e5-3494-4ff9-9272-316e9f1269c3
CSeq: 26366 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.12.0
Content-Length:  0


<--- Transmitting SIP request (1434 bytes) to UDP:54.172.60.0:5060 --->
INVITE sip:+919978747153@OM.pstn.twilio.com SIP/2.0
Via: SIP/2.0/UDP 94.94.94.94:8631;rport;branch=z9hG4bKPj5e7c1459-eece-471e-8a4c-facc2544b05e
From: <sip:1234567890@94.94.94.94>;tag=e5ba1d96-1c5e-4896-94e8-e2c0b406056e
To: <sip:+919978747153@OM.pstn.twilio.com>
Contact: <sip:asterisk@94.94.94.94:8631>
Call-ID: fe7639e5-3494-4ff9-9272-316e9f1269c3
CSeq: 26367 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.12.0
Proxy-Authorization: Digest username="logicsadmin", realm="sip.twilio.com", nonce="pHZpzI0OjeXtRh6ZN0BHNiwHXk2v45-1XTFjt0ZVkU29UnDK", uri="sip:+919978747153@OM.pstn.twilio.com", response="7080ebab803e00764803ad8cfdf693d4", cnonce="12e1d084f6e14c7eb66747f5e3f846d3", opaque="0d975d0cc5e2e4777859904b9029d3aa", qop=auth, nc=00000001
P-Asserted-Identity: <sip:1234567890@94.94.94.94>
Remote-Party-ID: <sip:1234567890@94.94.94.94>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 1841926769 1841926769 IN IP4 94.94.94.94
s=Asterisk
c=IN IP4 94.94.94.94
t=0 0
m=audio 15320 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (419 bytes) from UDP:54.172.60.0:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 94.94.94.94:8631;rport=8631;branch=z9hG4bKPj5e7c1459-eece-471e-8a4c-facc2544b05e;received=94.94.94.94
From: <sip:1234567890@94.94.94.94>;tag=e5ba1d96-1c5e-4896-94e8-e2c0b406056e
To: <sip:+919978747153@OM.pstn.twilio.com>
Call-ID: fe7639e5-3494-4ff9-9272-316e9f1269c3
CSeq: 26367 INVITE
Server: Twilio Gateway
Content-Length: 0


<--- Received SIP response (858 bytes) from UDP:54.172.60.0:5060 --->
SIP/2.0 183 Session progress
CSeq: 26367 INVITE
Call-ID: fe7639e5-3494-4ff9-9272-316e9f1269c3
From: <sip:1234567890@94.94.94.94>;tag=e5ba1d96-1c5e-4896-94e8-e2c0b406056e
To: <sip:+919978747153@OM.pstn.twilio.com>;tag=38145404_c3356d0b_310128f5-ceca-451c-88bd-190416bd1457
Via: SIP/2.0/UDP 94.94.94.94:8631;received=94.94.94.94;rport=8631;branch=z9hG4bKPj5e7c1459-eece-471e-8a4c-facc2544b05e
Record-Route: <sip:54.172.60.0;lr>
Server: Twilio
Contact: <sip:172.25.19.63:5060>
Content-Type: application/sdp
X-Twilio-CallSid: CAb53d97dcb53a05c31d63f5489bd2c6df
Content-Length: 253

v=0
o=root 806957316 806957316 IN IP4 172.18.168.178
s=Twilio Media Gateway
c=IN IP4 168.86.137.73
t=0 0
m=audio 14780 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

    -- PJSIP/twilio_outbound-00000013 is making progress passing it to PJSIP/1234567890-00000012
    -- Call on PJSIP/twilio_outbound-00000013 placed on hold
<--- Transmitting SIP response (1230 bytes) to WSS:49.49.49.49:40531 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS ffuq9ma9p2hg.invalid;rport=40531;received=49.49.49.49;branch=z9hG4bK8601973
Call-ID: kdogmfmrbr2kr08m53ps
From: <sip:1234567890@OMMS.sytes.net>;tag=ol1746ig2s
To: <sip:1234567890@OMMS.sytes.net>;tag=cf9161cd-fa80-4cc6-b9a6-a52f1eaeb28a
CSeq: 8822 INVITE
Server: Asterisk PBX 20.12.0
Contact: <sip:94.94.94.94:8089;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   657

v=0
o=- 1960213914208641035 4 IN IP4 94.94.94.94
s=Asterisk
c=IN IP4 94.94.94.94
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 13044 UDP/TLS/RTP/SAVPF 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 B3:BE:C0:7B:48:58:E8:E2:CB:F2:43:2C:28:59:C6:71:72:B3:5C:AE:03:31:64:E7:A5:13:6A:37:72:9B:F8:AE
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=recvonly
a=rtcp-mux
a=ssrc:328122174 cname:dea7f4a1-8e78-413f-8e09-735ddfde9046
a=msid:66ca7cf7-25c0-49e3-8a98-cb984c8820f4 5b2983d9-8ece-469b-a695-249f4965eedf
a=rtcp-fb:* transport-cc
a=mid:0

    -- Started music on hold, class 'default', on channel 'PJSIP/twilio_outbound-00000013'
    -- PJSIP/twilio_outbound-00000013 requested media update control 26, passing it to PJSIP/1234567890-00000012
<--- Received SIP response (563 bytes) from UDP:54.172.60.0:5060 --->
SIP/2.0 180 Ringing
CSeq: 26367 INVITE
Call-ID: fe7639e5-3494-4ff9-9272-316e9f1269c3
From: <sip:1234567890@94.94.94.94>;tag=e5ba1d96-1c5e-4896-94e8-e2c0b406056e
To: <sip:+919978747153@OM.pstn.twilio.com>;tag=38145404_c3356d0b_310128f5-ceca-451c-88bd-190416bd1457
Via: SIP/2.0/UDP 94.94.94.94:8631;received=94.94.94.94;rport=8631;branch=z9hG4bKPj5e7c1459-eece-471e-8a4c-facc2544b05e
Record-Route: <sip:54.172.60.0;lr>
Server: Twilio
Contact: <sip:172.25.19.63:5060>
X-Twilio-CallSid: CAb53d97dcb53a05c31d63f5489bd2c6df
Content-Length: 0


    -- PJSIP/twilio_outbound-00000013 is ringing
<--- Transmitting SIP response (1230 bytes) to WSS:49.49.49.49:40531 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS ffuq9ma9p2hg.invalid;rport=40531;received=49.49.49.49;branch=z9hG4bK8601973
Call-ID: kdogmfmrbr2kr08m53ps
From: <sip:1234567890@OMMS.sytes.net>;tag=ol1746ig2s
To: <sip:1234567890@OMMS.sytes.net>;tag=cf9161cd-fa80-4cc6-b9a6-a52f1eaeb28a
CSeq: 8822 INVITE
Server: Asterisk PBX 20.12.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:94.94.94.94:8089;transport=ws>
Content-Type: application/sdp
Content-Length:   657

v=0
o=- 1960213914208641035 4 IN IP4 94.94.94.94
s=Asterisk
c=IN IP4 94.94.94.94
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 13044 UDP/TLS/RTP/SAVPF 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 B3:BE:C0:7B:48:58:E8:E2:CB:F2:43:2C:28:59:C6:71:72:B3:5C:AE:03:31:64:E7:A5:13:6A:37:72:9B:F8:AE
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=recvonly
a=rtcp-mux
a=ssrc:328122174 cname:dea7f4a1-8e78-413f-8e09-735ddfde9046
a=msid:66ca7cf7-25c0-49e3-8a98-cb984c8820f4 5b2983d9-8ece-469b-a695-249f4965eedf
a=rtcp-fb:* transport-cc
a=mid:0

<--- Received SIP response (895 bytes) from UDP:54.172.60.0:5060 --->
SIP/2.0 200 OK
CSeq: 26367 INVITE
Call-ID: fe7639e5-3494-4ff9-9272-316e9f1269c3
From: <sip:1234567890@94.94.94.94>;tag=e5ba1d96-1c5e-4896-94e8-e2c0b406056e
To: <sip:+919978747153@OM.pstn.twilio.com>;tag=38145404_c3356d0b_310128f5-ceca-451c-88bd-190416bd1457
Via: SIP/2.0/UDP 94.94.94.94:8631;received=94.94.94.94;rport=8631;branch=z9hG4bKPj5e7c1459-eece-471e-8a4c-facc2544b05e
Record-Route: <sip:54.172.60.0;lr>
Server: Twilio
Contact: <sip:172.25.19.63:5060>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY
Content-Type: application/sdp
X-Twilio-CallSid: CAb53d97dcb53a05c31d63f5489bd2c6df
Content-Length: 253

v=0
o=root 806957316 806957316 IN IP4 172.18.168.178
s=Twilio Media Gateway
c=IN IP4 168.86.137.73
t=0 0
m=audio 14780 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (493 bytes) to UDP:54.172.60.0:5060 --->
ACK sip:172.25.19.63:5060 SIP/2.0
Via: SIP/2.0/UDP 94.94.94.94:8631;rport;branch=z9hG4bKPj24f711c6-768c-46bc-b187-71abb772bf44
From: <sip:1234567890@94.94.94.94>;tag=e5ba1d96-1c5e-4896-94e8-e2c0b406056e
To: <sip:+919978747153@OM.pstn.twilio.com>;tag=38145404_c3356d0b_310128f5-ceca-451c-88bd-190416bd1457
Call-ID: fe7639e5-3494-4ff9-9272-316e9f1269c3
CSeq: 26367 ACK
Route: <sip:54.172.60.0:5060;lr>
Max-Forwards: 70
User-Agent: Asterisk PBX 20.12.0
Content-Length:  0


    -- PJSIP/twilio_outbound-00000013 answered PJSIP/1234567890-00000012
<--- Transmitting SIP response (1264 bytes) to WSS:49.49.49.49:40531 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS ffuq9ma9p2hg.invalid;rport=40531;received=49.49.49.49;branch=z9hG4bK8601973
Call-ID: kdogmfmrbr2kr08m53ps
From: <sip:1234567890@OMMS.sytes.net>;tag=ol1746ig2s
To: <sip:1234567890@OMMS.sytes.net>;tag=cf9161cd-fa80-4cc6-b9a6-a52f1eaeb28a
CSeq: 8822 INVITE
Server: Asterisk PBX 20.12.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:94.94.94.94:8089;transport=ws>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   657

v=0
o=- 1960213914208641035 4 IN IP4 94.94.94.94
s=Asterisk
c=IN IP4 94.94.94.94
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 13044 UDP/TLS/RTP/SAVPF 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 B3:BE:C0:7B:48:58:E8:E2:CB:F2:43:2C:28:59:C6:71:72:B3:5C:AE:03:31:64:E7:A5:13:6A:37:72:9B:F8:AE
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=recvonly
a=rtcp-mux
a=ssrc:328122174 cname:dea7f4a1-8e78-413f-8e09-735ddfde9046
a=msid:66ca7cf7-25c0-49e3-8a98-cb984c8820f4 5b2983d9-8ece-469b-a695-249f4965eedf
a=rtcp-fb:* transport-cc
a=mid:0

    -- Channel PJSIP/twilio_outbound-00000013 joined 'simple_bridge' basic-bridge <2513e53a-6538-4361-8336-668fbbf2b142>
    -- Channel PJSIP/1234567890-00000012 joined 'simple_bridge' basic-bridge <2513e53a-6538-4361-8336-668fbbf2b142>
    -- Stopped music on hold on PJSIP/twilio_outbound-00000013
<--- Received SIP request (390 bytes) from WSS:49.49.49.49:40531 --->
ACK sip:94.94.94.94:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS ffuq9ma9p2hg.invalid;branch=z9hG4bK9941581
Max-Forwards: 70
To: <sip:1234567890@OMMS.sytes.net>;tag=cf9161cd-fa80-4cc6-b9a6-a52f1eaeb28a
From: <sip:1234567890@OMMS.sytes.net>;tag=ol1746ig2s
Call-ID: kdogmfmrbr2kr08m53ps
CSeq: 8822 ACK
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Length: 0


<--- Received SIP request (440 bytes) from WSS:49.49.49.49:40531 --->
BYE sip:94.94.94.94:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS ffuq9ma9p2hg.invalid;branch=z9hG4bK1274605
Max-Forwards: 70
To: <sip:1234567890@OMMS.sytes.net>;tag=cf9161cd-fa80-4cc6-b9a6-a52f1eaeb28a
From: <sip:1234567890@OMMS.sytes.net>;tag=ol1746ig2s
Call-ID: kdogmfmrbr2kr08m53ps
CSeq: 8823 BYE
Reason: SIP;cause=488;text="Not Acceptable Here"
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Length: 0


<--- Transmitting SIP response (357 bytes) to WSS:49.49.49.49:40531 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS ffuq9ma9p2hg.invalid;rport=40531;received=49.49.49.49;branch=z9hG4bK1274605
Call-ID: kdogmfmrbr2kr08m53ps
From: <sip:1234567890@OMMS.sytes.net>;tag=ol1746ig2s
To: <sip:1234567890@OMMS.sytes.net>;tag=cf9161cd-fa80-4cc6-b9a6-a52f1eaeb28a
CSeq: 8823 BYE
Server: Asterisk PBX 20.12.0
Content-Length:  0


    -- Channel PJSIP/1234567890-00000012 left 'simple_bridge' basic-bridge <2513e53a-6538-4361-8336-668fbbf2b142>
    -- Channel PJSIP/twilio_outbound-00000013 left 'simple_bridge' basic-bridge <2513e53a-6538-4361-8336-668fbbf2b142>
  == Spawn extension (from-internal, 1234567890, 1) exited non-zero on 'PJSIP/1234567890-00000012'
<--- Transmitting SIP request (517 bytes) to UDP:54.172.60.0:5060 --->
BYE sip:172.25.19.63:5060 SIP/2.0
Via: SIP/2.0/UDP 94.94.94.94:8631;rport;branch=z9hG4bKPjbcf4e230-100e-4f6f-a0c3-8b629a38feb5
From: <sip:1234567890@94.94.94.94>;tag=e5ba1d96-1c5e-4896-94e8-e2c0b406056e
To: <sip:+919978747153@OM.pstn.twilio.com>;tag=38145404_c3356d0b_310128f5-ceca-451c-88bd-190416bd1457
Call-ID: fe7639e5-3494-4ff9-9272-316e9f1269c3
CSeq: 26368 BYE
Route: <sip:54.172.60.0:5060;lr>
Reason: Q.850;cause=58
Max-Forwards: 70
User-Agent: Asterisk PBX 20.12.0
Content-Length:  0


<--- Received SIP response (485 bytes) from UDP:54.172.60.0:5060 --->
SIP/2.0 200 OK
CSeq: 26368 BYE
Call-ID: fe7639e5-3494-4ff9-9272-316e9f1269c3
From: <sip:1234567890@94.94.94.94>;tag=e5ba1d96-1c5e-4896-94e8-e2c0b406056e
To: <sip:+919978747153@OM.pstn.twilio.com>;tag=38145404_c3356d0b_310128f5-ceca-451c-88bd-190416bd1457
Via: SIP/2.0/UDP 94.94.94.94:8631;received=94.94.94.94;rport=8631;branch=z9hG4bKPjbcf4e230-100e-4f6f-a0c3-8b629a38feb5
Server: Twilio
X-Twilio-CallSid: CAb53d97dcb53a05c31d63f5489bd2c6df
Content-Length: 0

thanks and regards

4

The WebRTC client did not like the 200 OK. There are no ICE candidates which could be the reason. It is also recvonly which it may not like (not sure why). Examining the WebRTC client side may answer what it dislikes to confirm.

What is the contents of rtp.conf?
Where is this running?

5

many thanks again for your reply. please find my contents for rtp.conf

 cat /etc/asterisk/rtp.conf
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000
stunaddr=stun.l.google.com:19302
icesupport=yes

*CLI> rtp show settings


General Settings:
----------------
  Port start:      10000
  Port end:        20000
  Checksums:       Yes
  DTMF Timeout:    1200
  Strict RTP:      Yes
  Probation:       4 frames
  Replay Protect:  Yes
  ICE support:     Yes
  STUN address:    74.128.125.71:19302

**Actual working scenario is : **

i have Webphone which uses webrtc (port 8089)
When opened Webphone, it looks something like below screenshot

image
image1745×1145 45.7 KB

  1. User registration happens successfully via Webphone
  2. Once registered, user dials to 1234567890 then call will be dialed and should be to bridged 919978747153. In the call both caller and callee should be on the call itself without disconnecting it.

please help me with this

thanks and regards

6

@jcolp

i also tested Webrtc internal calls. calls getting dropped. This one iam not using twilio anywhere

CLI>
<--- Transmitting SIP request (465 bytes) to WSS:49.49.49.49:40910 --->
OPTIONS sip:ne9iqd53@49.49.49.49:40910;transport=ws SIP/2.0
Via: SIP/2.0/WSS 94.94.94.94:8089;rport;branch=z9hG4bKPjb950ea54-46a3-4245-90a8-3f52841f4966;alias
From: <sip:101@localhost>;tag=42c8c197-6ba0-4536-bac4-94bab20697b1
To: <sip:ne9iqd53@49.49.49.49>
Contact: <sip:101@localhost:5060;transport=ws>
Call-ID: ca4459af-0b4d-4053-9447-75d36ff0ce08
CSeq: 39818 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.12.0
Content-Length:  0


<--- Received SIP response (488 bytes) from WSS:49.49.49.49:40910 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 94.94.94.94:8089;rport;branch=z9hG4bKPjb950ea54-46a3-4245-90a8-3f52841f4966;alias
To: <sip:ne9iqd53@49.49.49.49>;tag=c3unfd78d7
From: <sip:101@localhost>;tag=42c8c197-6ba0-4536-bac4-94bab20697b1
Call-ID: ca4459af-0b4d-4053-9447-75d36ff0ce08
CSeq: 39818 OPTIONS
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Accept: application/sdp,application/dtmf-relay
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Length: 0


[Feb 18 13:56:09] NOTICE[6590]: res_pjsip/pjsip_transport_management.c:170 idle_sched_cb: Shutting down transport 'WSS to 49.49.49.49:39835' since no request was received in 32 seconds
[Feb 18 13:56:09] ERROR[6645]: res_http_websocket.c:567 ws_safe_read: Error reading from web socket: Broken pipe
[Feb 18 13:56:09] ERROR[6582]: iostream.c:563 ast_iostream_close: SSL_shutdown() failed: error:00000001:lib(0)::reason(1), Internal SSL error
  == WebSocket connection from '49.49.49.49:39835' closed
  == WebSocket connection from '49.49.49.49:40318' for protocol 'sip' accepted using version '13'
<--- Received SIP request (1786 bytes) from WSS:49.49.49.49:39687 --->
INVITE sip:101@OM.sytes.net SIP/2.0
Via: SIP/2.0/WSS c89cauj91p9n.invalid;branch=z9hG4bK963066
Max-Forwards: 70
To: <sip:101@OM.sytes.net>
From: <sip:102@OM.sytes.net>;tag=66b5lrrhgs
Call-ID: dsuf2l5oh9odrfgdd5d0
CSeq: 4624 INVITE
Contact: <sip:5h0jbt8v@c89cauj91p9n.invalid;transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Type: application/sdp
Content-Length: 1295

v=0
o=- 1113747485962159314 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 3f4fba65-e384-4fd5-a516-8a9a09fc4982
m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:dWqy
a=ice-pwd:wO/LpGtYtKocJVD8YQ3tUQ2d
a=ice-options:trickle
a=fingerprint:sha-256 C1:C6:9F:F4:8E:A8:02:25:85:2B:E3:38:79:60:E0:E5:BD:86:FF:5B:48:1F:EF:72:07:BD:E2:54:88:20:5B:F2
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:3f4fba65-e384-4fd5-a516-8a9a09fc4982 67329147-41fe-483a-b8ac-97e3ee9b8a8f
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:2736148472 cname:fSfMV+xrIT4DiDlu
a=ssrc:2736148472 msid:3f4fba65-e384-4fd5-a516-8a9a09fc4982 67329147-41fe-483a-b8ac-97e3ee9b8a8f

<--- Transmitting SIP response (476 bytes) to WSS:49.49.49.49:39687 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS c89cauj91p9n.invalid;rport=39687;received=49.49.49.49;branch=z9hG4bK963066
Call-ID: dsuf2l5oh9odrfgdd5d0
From: <sip:102@OM.sytes.net>;tag=66b5lrrhgs
To: <sip:101@OM.sytes.net>;tag=z9hG4bK963066
CSeq: 4624 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1739883371/13c4efbd25222d141ed14cf7556fe6ab",opaque="4154a82f59dd60d8",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.12.0
Content-Length:  0


<--- Received SIP request (343 bytes) from WSS:49.49.49.49:39687 --->
ACK sip:101@OM.sytes.net SIP/2.0
Via: SIP/2.0/WSS c89cauj91p9n.invalid;branch=z9hG4bK743963
Max-Forwards: 70
To: <sip:101@OM.sytes.net>;tag=z9hG4bK963066
From: <sip:102@OM.sytes.net>;tag=66b5lrrhgs
Call-ID: dsuf2l5oh9odrfgdd5d0
CSeq: 4624 ACK
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Length: 0


<--- Received SIP request (2065 bytes) from WSS:49.49.49.49:39687 --->
INVITE sip:101@OM.sytes.net SIP/2.0
Via: SIP/2.0/WSS c89cauj91p9n.invalid;branch=z9hG4bK8286834
Max-Forwards: 70
To: <sip:101@OM.sytes.net>
From: <sip:102@OM.sytes.net>;tag=66b5lrrhgs
Call-ID: dsuf2l5oh9odrfgdd5d0
CSeq: 4625 INVITE
Authorization: Digest algorithm=MD5, username="102", realm="asterisk", nonce="1739883371/13c4efbd25222d141ed14cf7556fe6ab", uri="sip:101@OM.sytes.net", response="28147f87c2a5b6741a85136e520d138d", opaque="4154a82f59dd60d8", qop=auth, cnonce="bekqnahp6lph", nc=00000001
Contact: <sip:5h0jbt8v@c89cauj91p9n.invalid;transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Type: application/sdp
Content-Length: 1295

v=0
o=- 1113747485962159314 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 3f4fba65-e384-4fd5-a516-8a9a09fc4982
m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:dWqy
a=ice-pwd:wO/LpGtYtKocJVD8YQ3tUQ2d
a=ice-options:trickle
a=fingerprint:sha-256 C1:C6:9F:F4:8E:A8:02:25:85:2B:E3:38:79:60:E0:E5:BD:86:FF:5B:48:1F:EF:72:07:BD:E2:54:88:20:5B:F2
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:3f4fba65-e384-4fd5-a516-8a9a09fc4982 67329147-41fe-483a-b8ac-97e3ee9b8a8f
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:2736148472 cname:fSfMV+xrIT4DiDlu
a=ssrc:2736148472 msid:3f4fba65-e384-4fd5-a516-8a9a09fc4982 67329147-41fe-483a-b8ac-97e3ee9b8a8f

<--- Transmitting SIP response (307 bytes) to WSS:49.49.49.49:39687 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS c89cauj91p9n.invalid;rport=39687;received=49.49.49.49;branch=z9hG4bK8286834
Call-ID: dsuf2l5oh9odrfgdd5d0
From: <sip:102@OM.sytes.net>;tag=66b5lrrhgs
To: <sip:101@OM.sytes.net>
CSeq: 4625 INVITE
Server: Asterisk PBX 20.12.0
Content-Length:  0


    -- Executing [101@from-internal:1] Dial("PJSIP/102-00000000", "PJSIP/101") in new stack
    -- Called PJSIP/101
<--- Transmitting SIP request (1669 bytes) to WSS:49.49.49.49:40910 --->
INVITE sip:ne9iqd53@49.49.49.49:40910;transport=ws SIP/2.0
Via: SIP/2.0/WSS 94.94.94.94:8089;rport;branch=z9hG4bKPj21a8697a-13ae-40e1-a82c-26a9dcc36594;alias
From: <sip:102@localhost>;tag=693aaca1-258f-479b-a9bf-f3e25397d082
To: <sip:ne9iqd53@49.49.49.49>
Contact: <sip:asterisk@localhost:5060;transport=ws>
Call-ID: 2f181de2-fc65-4d9f-9d5c-08a6128d90db
CSeq: 13685 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.12.0
Content-Type: application/sdp
Content-Length:   957

v=0
o=- 1661768713 1661768713 IN IP4 94.94.94.94
s=Asterisk
c=IN IP4 94.94.94.94
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 19450 UDP/TLS/RTP/SAVPF 0 8 18 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 CB:97:08:03:C4:FD:8C:A0:C5:F9:31:48:F5:38:D4:A0:1A:BB:10:D5:1E:E2:53:10:03:D1:1E:A4:17:B6:52:67
a=ice-ufrag:021eb5aa39988f9808f338971e6333ac
a=ice-pwd:22ca096215a7f71e02d3e9c62c64a9ee
a=candidate:H5eb1a03b 1 UDP 2130706431 94.94.94.94 19450 typ host
a=candidate:He9960a21 1 UDP 2130706431 fe80::777f:e4b:cc09:8778 19450 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtcp-mux
a=ssrc:1102308645 cname:c1a229a4-648f-4bb0-b4f5-f8bbfcecea15
a=msid:854b589f-5cda-4dac-8579-b38289b75ef8 5d1afee1-74b3-45ce-a201-27a2028479a9
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Received SIP response (364 bytes) from WSS:49.49.49.49:40910 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 94.94.94.94:8089;rport;branch=z9hG4bKPj21a8697a-13ae-40e1-a82c-26a9dcc36594;alias
To: <sip:ne9iqd53@49.49.49.49>
From: <sip:102@localhost>;tag=693aaca1-258f-479b-a9bf-f3e25397d082
Call-ID: 2f181de2-fc65-4d9f-9d5c-08a6128d90db
CSeq: 13685 INVITE
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Length: 0


<--- Received SIP response (439 bytes) from WSS:49.49.49.49:40910 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 94.94.94.94:8089;rport;branch=z9hG4bKPj21a8697a-13ae-40e1-a82c-26a9dcc36594;alias
To: <sip:ne9iqd53@49.49.49.49>;tag=27l36iirt3
From: <sip:102@localhost>;tag=693aaca1-258f-479b-a9bf-f3e25397d082
Call-ID: 2f181de2-fc65-4d9f-9d5c-08a6128d90db
CSeq: 13685 INVITE
Contact: <sip:ne9iqd53@ggb30vb1664o.invalid;transport=ws>
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Length: 0


    -- PJSIP/101-00000001 is ringing
<--- Transmitting SIP response (514 bytes) to WSS:49.49.49.49:39687 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS c89cauj91p9n.invalid;rport=39687;received=49.49.49.49;branch=z9hG4bK8286834
Call-ID: dsuf2l5oh9odrfgdd5d0
From: <sip:102@OM.sytes.net>;tag=66b5lrrhgs
To: <sip:101@OM.sytes.net>;tag=1056a3b7-dedf-4196-8a24-e9dfa85aa60c
CSeq: 4625 INVITE
Server: Asterisk PBX 20.12.0
Contact: <sip:94.94.94.94:8089;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Content-Length:  0


<--- Transmitting SIP request (465 bytes) to WSS:49.49.49.49:39687 --->
OPTIONS sip:5h0jbt8v@49.49.49.49:39687;transport=ws SIP/2.0
Via: SIP/2.0/WSS 94.94.94.94:8089;rport;branch=z9hG4bKPj6d563cf1-0dc5-43d4-b73c-0f8a1682fc6a;alias
From: <sip:102@localhost>;tag=09b5089e-adb7-42cf-b2ef-c37d3e345f6c
To: <sip:5h0jbt8v@49.49.49.49>
Contact: <sip:102@localhost:5060;transport=ws>
Call-ID: 4d1247c6-cada-4bc4-a67f-bfc0995a38c7
CSeq: 42031 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.12.0
Content-Length:  0


<--- Received SIP response (488 bytes) from WSS:49.49.49.49:39687 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 94.94.94.94:8089;rport;branch=z9hG4bKPj6d563cf1-0dc5-43d4-b73c-0f8a1682fc6a;alias
To: <sip:5h0jbt8v@49.49.49.49>;tag=t30ub9sldl
From: <sip:102@localhost>;tag=09b5089e-adb7-42cf-b2ef-c37d3e345f6c
Call-ID: 4d1247c6-cada-4bc4-a67f-bfc0995a38c7
CSeq: 42031 OPTIONS
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Accept: application/sdp,application/dtmf-relay
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Length: 0


<--- Received SIP response (1209 bytes) from WSS:49.49.49.49:40910 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 94.94.94.94:8089;rport;branch=z9hG4bKPj21a8697a-13ae-40e1-a82c-26a9dcc36594;alias
To: <sip:ne9iqd53@49.49.49.49>;tag=27l36iirt3
From: <sip:102@localhost>;tag=693aaca1-258f-479b-a9bf-f3e25397d082
Call-ID: 2f181de2-fc65-4d9f-9d5c-08a6128d90db
CSeq: 13685 INVITE
Contact: <sip:ne9iqd53@ggb30vb1664o.invalid;transport=ws>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Type: application/sdp
Content-Length: 678

v=0
o=- 2158858916471146395 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio-0
a=msid-semantic: WMS 29047db6-bca8-404d-be5f-09822e9979d9
m=audio 9 UDP/TLS/RTP/SAVPF 0 8 101
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:4ozk
a=ice-pwd:4Pa/HQ6ED9bppY2ZkHAd6ZYT
a=ice-options:trickle
a=fingerprint:sha-256 41:5E:9F:C1:61:41:73:44:C8:32:49:C4:E7:27:78:9C:4B:FD:3F:E3:EF:C0:B4:8E:3F:43:79:94:48:B0:3C:96
a=setup:active
a=mid:audio-0
a=sendrecv
a=msid:29047db6-bca8-404d-be5f-09822e9979d9 33ad866e-f85d-4c5d-ae78-79994b3d8bf8
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:1725253789 cname:h9GVACBznpqyTTUz

    -- Call on PJSIP/101-00000001 placed on hold
    -- Started music on hold, class 'default', on channel 'PJSIP/102-00000000'
<--- Transmitting SIP request (420 bytes) to WSS:49.49.49.49:40910 --->
ACK sip:ne9iqd53@49.49.49.49:40910;transport=ws SIP/2.0
Via: SIP/2.0/WSS 94.94.94.94:8089;rport;branch=z9hG4bKPj4f54fc72-08de-4fab-8cbf-0baa3e947f20;alias
From: <sip:102@localhost>;tag=693aaca1-258f-479b-a9bf-f3e25397d082
To: <sip:ne9iqd53@49.49.49.49>;tag=27l36iirt3
Call-ID: 2f181de2-fc65-4d9f-9d5c-08a6128d90db
CSeq: 13685 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.12.0
Content-Length:  0


    -- PJSIP/101-00000001 answered PJSIP/102-00000000
    -- Stopped music on hold on PJSIP/102-00000000
<--- Transmitting SIP response (1248 bytes) to WSS:49.49.49.49:39687 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS c89cauj91p9n.invalid;rport=39687;received=49.49.49.49;branch=z9hG4bK8286834
Call-ID: dsuf2l5oh9odrfgdd5d0
From: <sip:102@OM.sytes.net>;tag=66b5lrrhgs
To: <sip:101@OM.sytes.net>;tag=1056a3b7-dedf-4196-8a24-e9dfa85aa60c
CSeq: 4625 INVITE
Server: Asterisk PBX 20.12.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:94.94.94.94:8089;transport=ws>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   657

v=0
o=- 1113747485962159314 4 IN IP4 94.94.94.94
s=Asterisk
c=IN IP4 94.94.94.94
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 18790 UDP/TLS/RTP/SAVPF 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 EB:0A:E8:0B:5F:61:59:13:3D:C5:D6:D4:5A:D8:BD:54:09:A9:46:EB:92:81:F3:8B:51:15:4A:29:FD:CA:0B:6F
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=recvonly
a=rtcp-mux
a=ssrc:753396307 cname:a4fb1a5b-c897-49d4-aa05-7d16061e4903
a=msid:60c7108d-46bc-4122-87c1-a0e944581cf4 31095052-c41b-4ca4-ac76-bbf3139d40a0
a=rtcp-fb:* transport-cc
a=mid:0

    -- Channel PJSIP/101-00000001 joined 'simple_bridge' basic-bridge <9b83d1fe-db26-4432-9424-52607ef780cc>
    -- Channel PJSIP/102-00000000 joined 'simple_bridge' basic-bridge <9b83d1fe-db26-4432-9424-52607ef780cc>
    -- Started music on hold, class 'default', on channel 'PJSIP/101-00000001'
<--- Received SIP request (374 bytes) from WSS:49.49.49.49:39687 --->
ACK sip:94.94.94.94:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS c89cauj91p9n.invalid;branch=z9hG4bK3761397
Max-Forwards: 70
To: <sip:101@OM.sytes.net>;tag=1056a3b7-dedf-4196-8a24-e9dfa85aa60c
From: <sip:102@OM.sytes.net>;tag=66b5lrrhgs
Call-ID: dsuf2l5oh9odrfgdd5d0
CSeq: 4625 ACK
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Length: 0


<--- Received SIP request (424 bytes) from WSS:49.49.49.49:39687 --->
BYE sip:94.94.94.94:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS c89cauj91p9n.invalid;branch=z9hG4bK3413952
Max-Forwards: 70
To: <sip:101@OM.sytes.net>;tag=1056a3b7-dedf-4196-8a24-e9dfa85aa60c
From: <sip:102@OM.sytes.net>;tag=66b5lrrhgs
Call-ID: dsuf2l5oh9odrfgdd5d0
CSeq: 4626 BYE
Reason: SIP;cause=488;text="Not Acceptable Here"
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Length: 0


<--- Transmitting SIP response (341 bytes) to WSS:49.49.49.49:39687 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS c89cauj91p9n.invalid;rport=39687;received=49.49.49.49;branch=z9hG4bK3413952
Call-ID: dsuf2l5oh9odrfgdd5d0
From: <sip:102@OM.sytes.net>;tag=66b5lrrhgs
To: <sip:101@OM.sytes.net>;tag=1056a3b7-dedf-4196-8a24-e9dfa85aa60c
CSeq: 4626 BYE
Server: Asterisk PBX 20.12.0
Content-Length:  0


    -- Channel PJSIP/102-00000000 left 'simple_bridge' basic-bridge <9b83d1fe-db26-4432-9424-52607ef780cc>
    -- Channel PJSIP/101-00000001 left 'simple_bridge' basic-bridge <9b83d1fe-db26-4432-9424-52607ef780cc>
  == Spawn extension (from-internal, 101, 1) exited non-zero on 'PJSIP/102-00000000'
    -- Stopped music on hold on PJSIP/101-00000001
<--- Transmitting SIP request (444 bytes) to WSS:49.49.49.49:40910 --->
BYE sip:ne9iqd53@49.49.49.49:40910;transport=ws SIP/2.0
Via: SIP/2.0/WSS 94.94.94.94:8089;rport;branch=z9hG4bKPj22057b9e-5fb8-420f-9870-341cc73dfc11;alias
From: <sip:102@localhost>;tag=693aaca1-258f-479b-a9bf-f3e25397d082
To: <sip:ne9iqd53@49.49.49.49>;tag=27l36iirt3
Call-ID: 2f181de2-fc65-4d9f-9d5c-08a6128d90db
CSeq: 13686 BYE
Reason: Q.850;cause=58
Max-Forwards: 70
User-Agent: Asterisk PBX 20.12.0
Content-Length:  0


<--- Received SIP response (372 bytes) from WSS:49.49.49.49:40910 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 94.94.94.94:8089;rport;branch=z9hG4bKPj22057b9e-5fb8-420f-9870-341cc73dfc11;alias
To: <sip:ne9iqd53@49.49.49.49>;tag=27l36iirt3
From: <sip:102@localhost>;tag=693aaca1-258f-479b-a9bf-f3e25397d082
Call-ID: 2f181de2-fc65-4d9f-9d5c-08a6128d90db
CSeq: 13686 BYE
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Length: 0


<--- Received SIP request (374 bytes) from WSS:49.49.49.49:39687 --->
BYE sip:94.94.94.94:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS c89cauj91p9n.invalid;branch=z9hG4bK5949984
Max-Forwards: 70
To: <sip:101@OM.sytes.net>;tag=1056a3b7-dedf-4196-8a24-e9dfa85aa60c
From: <sip:102@OM.sytes.net>;tag=66b5lrrhgs
Call-ID: dsuf2l5oh9odrfgdd5d0
CSeq: 4627 BYE
Supported: outbound
User-Agent: SIP.js/0.11.0
Content-Length: 0


<--- Transmitting SIP response (364 bytes) to WSS:49.49.49.49:39687 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/WSS c89cauj91p9n.invalid;rport;received=49.49.49.49;branch=z9hG4bK5949984
Call-ID: dsuf2l5oh9odrfgdd5d0
From: <sip:102@OM.sytes.net>;tag=66b5lrrhgs
To: <sip:101@OM.sytes.net>;tag=1056a3b7-dedf-4196-8a24-e9dfa85aa60c
CSeq: 4627 BYE
Server: Asterisk PBX 20.12.0
Content-Length:  0

Please help me with this
thanks and regards

7

I don’t have an immediate answer for this. The fact the c= line contains “0.0.0.0” could be the cause of the recvonly, though I would still expect ICE candidates to be present.

Did you build Asterisk yourself?

8

thanks for your reply @jcolp ,

should i need to configure ICE candidates in rtp.conf or should i leave it blank. please suggest

yes, i used below link to build asterisk

https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20-current.tar.gz

and i installed it on below system.

 ~]# 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"

~]# cat /etc/redhat-release
CentOS Stream release 9

~]# df -h  /
Filesystem              Size  Used Avail Use% Mounted on
/dev/mapper/vg0-lvroot   79G   20G   60G  25% /

~]# free -h
               total        used        free      shared  buff/cache   available
Mem:           3.6Gi       605Mi       2.8Gi        26Mi       410Mi       3.0Gi
Swap:          511Mi          0B       511Mi

Also there is no firewall configured in my system. Without firewall is it possible to do please suggest

thanks and regards

9

Dear team,
please help
thanks and regards

10

i rectified call bridging issue… thanks and regards team