No Audio on Both Side on AWS EC2 instance

VinayakVasuMV · May 21, 2024, 12:31pm

I have installed asterisk on EC2 and im using call file to generate calls and bridge two sip calls but after answering i cant hear any sound on both sides,i have opened 5060 and 10000-20000 on firewall policy

ens5: inet 172.31.8.106

externip=52.66.91.105
localnet=172.31.0.0/20

ip-172-31-8-106*CLI> !sudo php generateCall.php
    -- Attempting call on SIP/5000 for 1000@sip-connect:1 (Retry 1)
  == Using SIP RTP CoS mark 5
Reliably Transmitting (NAT) to 152.58.202.121:40202:
OPTIONS sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK59d3575e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as11d2fedc
To: <sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea>
Contact: <sip:asterisk@52.66.91.105:5065>
Call-ID: 24a80a2a7218537a305e38e66a65e33b@52.66.91.105:5065
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Audio is at 18772
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 152.58.202.121:40202:
INVITE sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK77279cbf;rport
Max-Forwards: 70
From: <sip:1234@52.66.91.105:5065>;tag=as3f12fa6e
To: <sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea>
Contact: <sip:1234@52.66.91.105:5065>
Call-ID: 5aba276f7721a88557aebdd7058d6b47@52.66.91.105:5065
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 491929673 491929673 IN IP4 52.66.91.105
s=Asterisk PBX 18.23.1
c=IN IP4 52.66.91.105
t=0 0
m=audio 18772 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv

---
    -- Called 5000
Retransmitting #1 (NAT) to 152.58.202.121:40202:
INVITE sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK77279cbf;rport
Max-Forwards: 70
From: <sip:1234@52.66.91.105:5065>;tag=as3f12fa6e
To: <sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea>
Contact: <sip:1234@52.66.91.105:5065>
Call-ID: 5aba276f7721a88557aebdd7058d6b47@52.66.91.105:5065
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 491929673 491929673 IN IP4 52.66.91.105
s=Asterisk PBX 18.23.1
c=IN IP4 52.66.91.105
t=0 0
m=audio 18772 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv

---

<--- SIP read from UDP:152.58.202.121:40202 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK59d3575e;rport=5065
Contact: <sip:152.58.202.121:40202>
To: <sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea>;tag=0e4a830c
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as11d2fedc
Call-ID: 24a80a2a7218537a305e38e66a65e33b@52.66.91.105:5065
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.4_1
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '24a80a2a7218537a305e38e66a65e33b@52.66.91.105:5065' Method: OPTIONS

<--- SIP read from UDP:152.58.202.121:40202 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK77279cbf;rport=5065
To: <sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea>
From: <sip:1234@52.66.91.105:5065>;tag=as3f12fa6e
Call-ID: 5aba276f7721a88557aebdd7058d6b47@52.66.91.105:5065
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Reliably Transmitting (NAT) to 100.64.216.4:5060:
OPTIONS sip:100.64.216.4 SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK434b4857;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as131a1948
To: <sip:100.64.216.4>
Contact: <sip:asterisk@52.66.91.105:5065>
Call-ID: 7c2657696cc8bad602b9c1ea278c8fb8@52.66.91.105:5065
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:152.58.202.121:40202 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK77279cbf;rport=5065
Contact: <sip:5000@152.58.202.121:40202;transport=UDP>
To: <sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea>;tag=173e7264
From: <sip:1234@52.66.91.105:5065>;tag=as3f12fa6e
Call-ID: 5aba276f7721a88557aebdd7058d6b47@52.66.91.105:5065
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.4_1
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:5000@152.58.202.121:40202;transport=UDP>
    -- SIP/5000-00000000 is ringing
Retransmitting #1 (NAT) to 100.64.216.4:5060:
OPTIONS sip:100.64.216.4 SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK434b4857;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as131a1948
To: <sip:100.64.216.4>
Contact: <sip:asterisk@52.66.91.105:5065>
Call-ID: 7c2657696cc8bad602b9c1ea278c8fb8@52.66.91.105:5065
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (NAT) to 100.64.216.4:5060:
OPTIONS sip:100.64.216.4 SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK434b4857;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as131a1948
To: <sip:100.64.216.4>
Contact: <sip:asterisk@52.66.91.105:5065>
Call-ID: 7c2657696cc8bad602b9c1ea278c8fb8@52.66.91.105:5065
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:152.58.202.121:40202 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK77279cbf;rport=5065
Contact: <sip:5000@152.58.202.121:40202;transport=UDP>
To: <sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea>;tag=173e7264
From: <sip:1234@52.66.91.105:5065>;tag=as3f12fa6e
Call-ID: 5aba276f7721a88557aebdd7058d6b47@52.66.91.105:5065
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.4_1
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 356

v=0
o=Zoiper 0 1902139426 IN IP4 152.58.202.121
s=Zoiper
c=IN IP4 152.58.202.121
t=0 0
m=audio 51751 RTP/AVP 0 106 9 8 3 101 98
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=sendrecv
a=rtcp-mux
<------------->
--- (13 headers 14 lines) ---
Got SDP version 1902139426 and unique parts [Zoiper 0 IN IP4 152.58.202.121]
Found RTP audio format 0
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found RTP audio format 98
Found audio description format opus for ID 106
Found audio description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 98
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7f7a18011290 -- Strict RTP learning after remote address set to: 152.58.202.121:51751
Peer audio RTP is at port 152.58.202.121:51751
sip_route_dump: route/path hop: <sip:5000@152.58.202.121:40202;transport=UDP>
Transmitting (NAT) to 152.58.202.121:40202:
ACK sip:5000@152.58.202.121:40202;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK067f5536;rport
Max-Forwards: 70
From: <sip:1234@52.66.91.105:5065>;tag=as3f12fa6e
To: <sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea>;tag=173e7264
Contact: <sip:1234@52.66.91.105:5065>
Call-ID: 5aba276f7721a88557aebdd7058d6b47@52.66.91.105:5065
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.23.1
Content-Length: 0


---
    -- SIP/5000-00000000 answered
    -- Executing [1000@sip-connect:1] Set("SIP/5000-00000000", "CHANNEL(hangup_handler_push)=hangup-handler-click2call,1000,1") in new stack
    -- Executing [1000@sip-connect:2] AGI("SIP/5000-00000000", "click2call_B") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/click2call_B
 click2call_B: [21-05-2024 12:18:21]  NOTICE[1716293898.0] [unknown] [1234] [DATABASE]:  Database connected successfully  (DNID  is unknown,  CALLERID  is 1234,  SOURCE  is ,  EXTENSION  is 1000)
 click2call_B: [21-05-2024 12:18:21]  NOTICE[1716293898.0] [unknown] [1234] []:  CLICK2CALL application is dialing for SIP/1000 for user ID   (DNID  is unknown,  CALLERID  is 1234,  SOURCE  is ,  EXTENSION  is 1000)
    -- AGI Script Executing Application: (Dial) Options: (SIP/1000,30,TtgU(answer-click2call-B))
  == Using SIP RTP CoS mark 5
Reliably Transmitting (NAT) to 152.58.202.121:34018:
OPTIONS sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1 SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK155dd388;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as3f0e2340
To: <sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1>
Contact: <sip:asterisk@52.66.91.105:5065>
Call-ID: 0c3f72792443b8be6738bfbf21cf9a9c@52.66.91.105:5065
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Audio is at 12258
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 152.58.202.121:34018:
INVITE sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1 SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK5189c73a;rport
Max-Forwards: 70
From: <sip:1234@52.66.91.105:5065>;tag=as3552f279
To: <sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1>
Contact: <sip:1234@52.66.91.105:5065>
Call-ID: 274b67001f889dae3346952d786c00b1@52.66.91.105:5065
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1889930371 1889930371 IN IP4 52.66.91.105
s=Asterisk PBX 18.23.1
c=IN IP4 52.66.91.105
t=0 0
m=audio 12258 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv

---
    -- Called SIP/1000
Retransmitting #3 (NAT) to 100.64.216.4:5060:
OPTIONS sip:100.64.216.4 SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK434b4857;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as131a1948
To: <sip:100.64.216.4>
Contact: <sip:asterisk@52.66.91.105:5065>
Call-ID: 7c2657696cc8bad602b9c1ea278c8fb8@52.66.91.105:5065
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to 152.58.202.121:34018:
INVITE sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1 SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK5189c73a;rport
Max-Forwards: 70
From: <sip:1234@52.66.91.105:5065>;tag=as3552f279
To: <sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1>
Contact: <sip:1234@52.66.91.105:5065>
Call-ID: 274b67001f889dae3346952d786c00b1@52.66.91.105:5065
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1889930371 1889930371 IN IP4 52.66.91.105
s=Asterisk PBX 18.23.1
c=IN IP4 52.66.91.105
t=0 0
m=audio 12258 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv

---

<--- SIP read from UDP:152.58.202.121:34018 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK155dd388;rport=5065
Contact: <sip:192.168.242.26:34018>
To: <sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1>;tag=f6dc6a65
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as3f0e2340
Call-ID: 0c3f72792443b8be6738bfbf21cf9a9c@52.66.91.105:5065
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.5.15 v2.10.19.5
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '0c3f72792443b8be6738bfbf21cf9a9c@52.66.91.105:5065' Method: OPTIONS

<--- SIP read from UDP:152.58.202.121:34018 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK5189c73a;rport=5065
To: <sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1>
From: <sip:1234@52.66.91.105:5065>;tag=as3552f279
Call-ID: 274b67001f889dae3346952d786c00b1@52.66.91.105:5065
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
       > 0x7f7a18011290 -- Strict RTP switching to RTP target address 152.58.202.121:51751 as source
Retransmitting #4 (NAT) to 100.64.216.4:5060:
OPTIONS sip:100.64.216.4 SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK434b4857;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as131a1948
To: <sip:100.64.216.4>
Contact: <sip:asterisk@52.66.91.105:5065>
Call-ID: 7c2657696cc8bad602b9c1ea278c8fb8@52.66.91.105:5065
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '7c2657696cc8bad602b9c1ea278c8fb8@52.66.91.105:5065' Method: OPTIONS
ip-172-31-8-106*CLI> 

<--- SIP read from UDP:152.58.202.121:34018 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK5189c73a;rport=5065
Contact: <sip:1000@152.58.202.121:34018;transport=UDP>
To: <sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1>;tag=0db5eb37
From: <sip:1234@52.66.91.105:5065>;tag=as3552f279
Call-ID: 274b67001f889dae3346952d786c00b1@52.66.91.105:5065
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.5.15 v2.10.19.5
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:1000@152.58.202.121:34018;transport=UDP>
    -- SIP/1000-00000001 is ringing
ip-172-31-8-106*CLI> 
ip-172-31-8-106*CLI> 
ip-172-31-8-106*CLI> 

<--- SIP read from UDP:152.58.202.121:34018 --->
REGISTER sip:52.66.91.105:5065;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.242.26:34018;branch=z9hG4bK-524287-1---5095481123235d40;rport
Max-Forwards: 70
Contact: <sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1>
To: <sip:1000@52.66.91.105:5065;transport=UDP>
From: <sip:1000@52.66.91.105:5065;transport=UDP>;tag=dc22534e
Call-ID: 4IsDSJL6Zeuyeif_QFSp9g..
CSeq: 9 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.5.15 v2.10.19.5
Authorization: Digest username="1000",realm="127.0.0.1",nonce="1065c139",uri="sip:52.66.91.105:5065;transport=UDP",response="8166d3a242955fd69bf39794f22422c5",algorithm=MD5
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to 152.58.202.121:34018 (NAT)
Sending to 152.58.202.121:34018 (NAT)

<--- Transmitting (NAT) to 152.58.202.121:34018 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.242.26:34018;branch=z9hG4bK-524287-1---5095481123235d40;received=152.58.202.121;rport=34018
From: <sip:1000@52.66.91.105:5065;transport=UDP>;tag=dc22534e
To: <sip:1000@52.66.91.105:5065;transport=UDP>;tag=as6d309f43
Call-ID: 4IsDSJL6Zeuyeif_QFSp9g..
CSeq: 9 REGISTER
Server: Asterisk PBX 18.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="127.0.0.1", nonce="216ab94a"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '4IsDSJL6Zeuyeif_QFSp9g..' in 32000 ms (Method: REGISTER)
ip-172-31-8-106*CLI> 
ip-172-31-8-106*CLI> 
ip-172-31-8-106*CLI> 
ip-172-31-8-106*CLI> 
ip-172-31-8-106*CLI> 

<--- SIP read from UDP:152.58.202.121:34018 --->
REGISTER sip:52.66.91.105:5065;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.242.26:34018;branch=z9hG4bK-524287-1---1c0688c649b1de3f;rport
Max-Forwards: 70
Contact: <sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1>
To: <sip:1000@52.66.91.105:5065;transport=UDP>
From: <sip:1000@52.66.91.105:5065;transport=UDP>;tag=dc22534e
Call-ID: 4IsDSJL6Zeuyeif_QFSp9g..
CSeq: 10 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.5.15 v2.10.19.5
Authorization: Digest username="1000",realm="127.0.0.1",nonce="216ab94a",uri="sip:52.66.91.105:5065;transport=UDP",response="6b33e4c7dfd8275331c37f0abc11197a",algorithm=MD5
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to 152.58.202.121:34018 (NAT)
Reliably Transmitting (NAT) to 152.58.202.121:34018:
OPTIONS sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1 SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK5ac8dad7;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as54264e92
To: <sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1>
Contact: <sip:asterisk@52.66.91.105:5065>
Call-ID: 293f478957edef3c404d5b3b716e0b5a@52.66.91.105:5065
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
ip-172-31-8-106*CLI> 

<--- Transmitting (NAT) to 152.58.202.121:34018 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.242.26:34018;branch=z9hG4bK-524287-1---1c0688c649b1de3f;received=152.58.202.121;rport=34018
From: <sip:1000@52.66.91.105:5065;transport=UDP>;tag=dc22534e
To: <sip:1000@52.66.91.105:5065;transport=UDP>;tag=as6d309f43
Call-ID: 4IsDSJL6Zeuyeif_QFSp9g..
CSeq: 10 REGISTER
Server: Asterisk PBX 18.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1>;expires=60
Date: Tue, 21 May 2024 12:18:23 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '4IsDSJL6Zeuyeif_QFSp9g..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:152.58.202.121:34018 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK5ac8dad7;rport=5065
Contact: <sip:192.168.242.26:34018>
To: <sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1>;tag=ccd67b13
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as54264e92
Call-ID: 293f478957edef3c404d5b3b716e0b5a@52.66.91.105:5065
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.5.15 v2.10.19.5
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '293f478957edef3c404d5b3b716e0b5a@52.66.91.105:5065' Method: OPTIONS

<--- SIP read from UDP:152.58.202.121:40202 --->


<------------->
       > 0x7f7a18011290 -- Strict RTP learning complete - Locking on source address 152.58.202.121:51751

<--- SIP read from UDP:152.58.202.121:34018 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK5189c73a;rport=5065
Contact: <sip:1000@152.58.202.121:34018;transport=UDP>
To: <sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1>;tag=0db5eb37
From: <sip:1234@52.66.91.105:5065>;tag=as3552f279
Call-ID: 274b67001f889dae3346952d786c00b1@52.66.91.105:5065
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.5.15 v2.10.19.5
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 344

v=0
o=Z 0 31083643 IN IP4 152.58.202.121
s=Z
c=IN IP4 152.58.202.121
t=0 0
m=audio 43976 RTP/AVP 0 106 9 8 3 101 98
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=sendrecv
a=rtcp-mux
<------------->
--- (13 headers 14 lines) ---
Got SDP version 31083643 and unique parts [Z 0 IN IP4 152.58.202.121]
Found RTP audio format 0
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found RTP audio format 98
Found audio description format opus for ID 106
Found audio description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 98
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7f7a18041a00 -- Strict RTP learning after remote address set to: 152.58.202.121:43976
Peer audio RTP is at port 152.58.202.121:43976
sip_route_dump: route/path hop: <sip:1000@152.58.202.121:34018;transport=UDP>
Transmitting (NAT) to 152.58.202.121:34018:
ACK sip:1000@152.58.202.121:34018;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK34832232;rport
Max-Forwards: 70
From: <sip:1234@52.66.91.105:5065>;tag=as3552f279
To: <sip:1000@152.58.202.121:34018;transport=UDP;rinstance=460a476607fe91d1>;tag=0db5eb37
Contact: <sip:1234@52.66.91.105:5065>
Call-ID: 274b67001f889dae3346952d786c00b1@52.66.91.105:5065
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.23.1
Content-Length: 0


---
    -- SIP/1000-00000001 answered SIP/5000-00000000
    -- SIP/1000-00000001 Internal Gosub(answer-click2call-B,s,1) start
    -- Executing [s@answer-click2call-B:1] AGI("SIP/1000-00000001", "answer_click2call") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/answer_click2call
       > 0x7f7a18041a00 -- Strict RTP switching to RTP target address 152.58.202.121:43976 as source
ip-172-31-8-106*CLI> 
ip-172-31-8-106*CLI> 
ip-172-31-8-106*CLI> 
ip-172-31-8-106*CLI> 
Reliably Transmitting (NAT) to 100.64.216.4:5060:
OPTIONS sip:100.64.216.4 SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK1d9b5e60;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as45bd4aef
To: <sip:100.64.216.4>
Contact: <sip:asterisk@52.66.91.105:5065>
Call-ID: 49f3b792490fbb0b5b0f0c607a9e9c05@52.66.91.105:5065
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
       > 0x7f7a18041a00 -- Strict RTP learning complete - Locking on source address 152.58.202.121:43976
Retransmitting #1 (NAT) to 100.64.216.4:5060:
OPTIONS sip:100.64.216.4 SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK1d9b5e60;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as45bd4aef
To: <sip:100.64.216.4>
Contact: <sip:asterisk@52.66.91.105:5065>
Call-ID: 49f3b792490fbb0b5b0f0c607a9e9c05@52.66.91.105:5065
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (NAT) to 100.64.216.4:5060:
OPTIONS sip:100.64.216.4 SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK1d9b5e60;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as45bd4aef
To: <sip:100.64.216.4>
Contact: <sip:asterisk@52.66.91.105:5065>
Call-ID: 49f3b792490fbb0b5b0f0c607a9e9c05@52.66.91.105:5065
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:152.58.202.121:40202 --->
REGISTER sip:52.66.91.105:5065;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 152.58.202.121:40202;branch=z9hG4bK-524287-1---29125b15c034e470;rport
Max-Forwards: 70
Contact: <sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea>
To: <sip:5000@52.66.91.105:5065;transport=UDP>
From: <sip:5000@52.66.91.105:5065;transport=UDP>;tag=2ff78f11
Call-ID: VffJ-Ccb1X2yhtmId6W1TQ..
CSeq: 9 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.4_1
Authorization: Digest username="5000",realm="127.0.0.1",nonce="290bdea9",uri="sip:52.66.91.105:5065;transport=UDP",response="e6940bd60fd746d19b93a2c996a86760",algorithm=MD5
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to 152.58.202.121:40202 (no NAT)
Sending to 152.58.202.121:40202 (no NAT)

<--- Transmitting (NAT) to 152.58.202.121:40202 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 152.58.202.121:40202;branch=z9hG4bK-524287-1---29125b15c034e470;received=152.58.202.121;rport=40202
From: <sip:5000@52.66.91.105:5065;transport=UDP>;tag=2ff78f11
To: <sip:5000@52.66.91.105:5065;transport=UDP>;tag=as42c94791
Call-ID: VffJ-Ccb1X2yhtmId6W1TQ..
CSeq: 9 REGISTER
Server: Asterisk PBX 18.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="127.0.0.1", nonce="581da599"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'VffJ-Ccb1X2yhtmId6W1TQ..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:152.58.202.121:40202 --->
REGISTER sip:52.66.91.105:5065;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 152.58.202.121:40202;branch=z9hG4bK-524287-1---38cb6ba849a52448;rport
Max-Forwards: 70
Contact: <sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea>
To: <sip:5000@52.66.91.105:5065;transport=UDP>
From: <sip:5000@52.66.91.105:5065;transport=UDP>;tag=2ff78f11
Call-ID: VffJ-Ccb1X2yhtmId6W1TQ..
CSeq: 10 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.4_1
Authorization: Digest username="5000",realm="127.0.0.1",nonce="581da599",uri="sip:52.66.91.105:5065;transport=UDP",response="367041575e2daf03a1d2c5b7a83c1006",algorithm=MD5
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to 152.58.202.121:40202 (no NAT)
Reliably Transmitting (NAT) to 152.58.202.121:40202:
OPTIONS sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea SIP/2.0
Via: SIP/2.0/UDP 52.66.91.105:5065;branch=z9hG4bK4ea84edd;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@52.66.91.105:5065>;tag=as4e743b5e
To: <sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea>
Contact: <sip:asterisk@52.66.91.105:5065>
Call-ID: 6b16a4d929ef3ceb007b7ef82d6d0c4c@52.66.91.105:5065
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 21 May 2024 12:18:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 152.58.202.121:40202 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 152.58.202.121:40202;branch=z9hG4bK-524287-1---38cb6ba849a52448;received=152.58.202.121;rport=40202
From: <sip:5000@52.66.91.105:5065;transport=UDP>;tag=2ff78f11
To: <sip:5000@52.66.91.105:5065;transport=UDP>;tag=as42c94791
Call-ID: VffJ-Ccb1X2yhtmId6W1TQ..
CSeq: 10 REGISTER
Server: Asterisk PBX 18.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:5000@152.58.202.121:40202;transport=UDP;rinstance=0921b16e7e339aea>;expires=60
Date: Tue, 21 May 2024 12:18:35 GMT
Content-Length: 0

jcolp · May 21, 2024, 12:36pm

The signaling has the correct IP address. If the remote side is behind NAT, then that has to be configured in sip.conf as well. If it is then you’d need to see if any traffic is being received using a packet capture, if none is then the problem is outside of Asterisk.

VinayakVasuMV · May 21, 2024, 12:52pm

Remote side is not behind NAT and i also tried nat=force_rport,comedia.
I can call the two sip extension and can hear both sides but when using call.file to dial these two extension then no audio
Also when the B party call is answer but still ringing is hear in the A party sip extension

VinayakVasuMV · May 21, 2024, 1:24pm

These message are shown after hangup

[May 21 13:23:32] WARNING[111010]: chan_sip.c:4420 __sip_autodestruct: Autodestruct on dialog '7df272983cbaa33a7c0f0c6a011cd7bb@52.66.91.105:5065' with owner SIP/1000-00000000 in place (Method: BYE). Rescheduling destruction for 10000 ms
[May 21 13:23:38] WARNING[111010]: chan_sip.c:4420 __sip_autodestruct: Autodestruct on dialog '7df272983cbaa33a7c0f0c6a011cd7bb@52.66.91.105:5065' with owner SIP/1000-00000000 in place (Method: BYE). Rescheduling destruction for 10000 ms
[May 21 13:23:45] WARNING[111010]: chan_sip.c:4420 __sip_autodestruct: Autodestruct on dialog '7df272983cbaa33a7c0f0c6a011cd7bb@52.66.91.105:5065' with owner SIP/1000-00000000 in place (Method: BYE). Rescheduling destruction for 10000 ms
[May 21 13:23:51] WARNING[111010]: chan_sip.c:4420 __sip_autodestruct: Autodestruct on dialog '7df272983cbaa33a7c0f0c6a011cd7bb@52.66.91.105:5065' with owner SIP/1000-00000000 in place (Method: BYE). Rescheduling destruction for 10000 ms

system · June 20, 2024, 1:25pm

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