Sip client registration ok , debug shows correct action based on dial plan, RTP ports open but no sound

lmon · May 12, 2020, 7:17pm

Hi
I have my sip client which register, but when I try for example to get extension name via *65, I get no sound.
I’m using a manual install of FreePBX ( even I don’t think it is changing anything on this config point ).

here is some CLI debug

no client connected, so I can not set debug to extension :

adc1*CLI> SIP SET DEBUG PEER 5555
Unable to get IP address of peer '5555'

Then I register my client

 -- Registered SIP '5555' at 37.170.156.93:52566
[2020-05-12 20:35:27] NOTICE[21959]: chan_sip.c:25008 handle_response_peerpoke: Peer '5555' is now Reachable. (119ms / 2000ms)

Then I call the *65 ( to get my extension ) :

adc1*CLI> SIP SET DEBUG PEER 5555
SIP Debugging Enabled for IP: [My Client IP]
adc1*CLI> *******
No such command '*******' (type 'core show help *******' for other possible commands)
adc1*CLI> ****
No such command '****' (type 'core show help ****' for other possible commands)
adc1*CLI> *****
No such command '*****' (type 'core show help *****' for other possible commands)
adc1*CLI> *******
No such command '*******' (type 'core show help *******' for other possible commands)
adc1*CLI> *****
No such command '*****' (type 'core show help *****' for other possible commands)

<--- SIP read from UDP:[My Client IP]:52566 --->
INVITE sip:*65@[FQDN of my Server ]:6116 SIP/2.0
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---42f0001b95815d6e;rport
Max-Forwards: 70
Contact: <sip:[email protected]:36571>
To: <sip:*65@[FQDN of my Server ]:6116>
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 1 INVITE
Session-Expires: 1800
Min-SE: 90
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: 398

v=0
o=- 0 1 IN IP4 192.168.0.250
s=-
c=IN IP4 10.115.236.181
t=0 0
m=audio 4086 RTP/AVP 3 102 0 8 9 120 101
a=rtpmap:3 GSM/8000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 17 lines) ---
Sending to [My Client IP]:52566 (NAT)
Sending to [My Client IP]:52566 (NAT)
Using INVITE request as basis request - UBKs1nTIc_v_ckIarWFcqw..
Found peer '5555' for '5555' from [My Client IP]:52566

<--- Reliably Transmitting (NAT) to [My Client IP]:52566 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---42f0001b95815d6e;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as650fda78
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 1 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5dbbdc85"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'UBKs1nTIc_v_ckIarWFcqw..' in 7616 ms (Method: INVITE)

<--- SIP read from UDP:[My Client IP]:52566 --->
ACK sip:*65@[FQDN of my Server ]:6116 SIP/2.0
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---42f0001b95815d6e;rport
Max-Forwards: 70
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as650fda78
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:[My Client IP]:52566 --->
INVITE sip:*65@[FQDN of my Server ]:6116 SIP/2.0
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;rport
Max-Forwards: 70
Contact: <sip:[email protected]:36571>
To: <sip:*65@[FQDN of my Server ]:6116>
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Session-Expires: 1800
Min-SE: 90
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
Authorization: Digest username="5555",realm="asterisk",nonce="5dbbdc85",uri="sip:*65@[FQDN of my Server ]:6116",response="0bda3cdf17b7a72f627228325b12bacf",algorithm=MD5
Content-Length: 398

v=0
o=- 0 1 IN IP4 192.168.0.250
s=-
c=IN IP4 10.115.236.181
t=0 0
m=audio 4086 RTP/AVP 3 102 0 8 9 120 101
a=rtpmap:3 GSM/8000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (16 headers 17 lines) ---
Sending to [My Client IP]:52566 (NAT)
Using INVITE request as basis request - UBKs1nTIc_v_ckIarWFcqw..
Found peer '5555' for '5555' from [My Client IP]:52566
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Got SDP version 1 and unique parts [- 0 IN IP4 192.168.0.250]
Found RTP audio format 3
Found RTP audio format 102
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 120
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 102
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format opus for ID 120
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|gsm|alaw|g722|ilbc|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x559b9dbdfe40 -- Strict RTP learning after remote address set to: 10.115.236.181:4086
Peer audio RTP is at port 10.115.236.181:4086
Looking for *65 in from-internal (domain [FQDN of my Server ])
sip_route_dump: route/path hop: <sip:[email protected]:36571>

<--- Transmitting (NAT) to [My Client IP]:52566 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
Content-Length: 0


<------------>
    -- Executing [*65@from-internal:1] Set("SIP/5555-00000027", "CONNECTEDLINE(name-charset,i)=utf8") in new stack
    -- Executing [*65@from-internal:2] Set("SIP/5555-00000027", "CONNECTEDLINE(name,i)=Speak Extension") in new stack
    -- Executing [*65@from-internal:3] Set("SIP/5555-00000027", "CONNECTEDLINE(num,i)=*65") in new stack
    -- Executing [*65@from-internal:4] Answer("SIP/5555-00000027", "") in new stack
Audio is at 16602
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to [My Client IP]:52566 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
Retransmitting #1 (NAT) to [My Client IP]:52566:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
Retransmitting #2 (NAT) to [My Client IP]:52566:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Executing [*65@from-internal:5] Wait("SIP/5555-00000027", "1") in new stack
Retransmitting #3 (NAT) to [My Client IP]:52566:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:[My Client IP]:52566 --->


<------------->
    -- Executing [*65@from-internal:6] Macro("SIP/5555-00000027", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/5555-00000027", "TOUCH_MONITOR=1589308555.39") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/5555-00000027", "AMPUSER=5555") in new stack
    -- Executing [s@macro-user-callerid:3] Set("SIP/5555-00000027", "HOTDESCKCHAN=5555-00000027") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/5555-00000027", "HOTDESKEXTEN=5555") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/5555-00000027", "HOTDESKCALL=0") in new stack
    -- Executing [s@macro-user-callerid:6] ExecIf("SIP/5555-00000027", "0?Set(HOTDESKCALL=1)") in new stack
    -- Executing [s@macro-user-callerid:7] ExecIf("SIP/5555-00000027", "0?Set(CALLERID(name)=)") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("SIP/5555-00000027", "0?report") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/5555-00000027", "1?Set(REALCALLERIDNUM=5555)") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/5555-00000027", "AMPUSER=5555") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/5555-00000027", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:12] Set("SIP/5555-00000027", "AMPUSERCIDNAME=5555") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("SIP/5555-00000027", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("SIP/5555-00000027", "0?report") in new stack
    -- Executing [s@macro-user-callerid:15] Set("SIP/5555-00000027", "AMPUSERCID=5555") in new stack
    -- Executing [s@macro-user-callerid:16] Set("SIP/5555-00000027", "__DIAL_OPTIONS=HhTtr") in new stack
    -- Executing [s@macro-user-callerid:17] Set("SIP/5555-00000027", "CALLERID(all)="5555" <5555>") in new stack
    -- Executing [s@macro-user-callerid:18] ExecIf("SIP/5555-00000027", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-user-callerid:19] GotoIf("SIP/5555-00000027", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:20] ExecIf("SIP/5555-00000027", "0?Set(GROUP(concurrency_limit)=5555)") in new stack
    -- Executing [s@macro-user-callerid:21] ExecIf("SIP/5555-00000027", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:22] NoOp("SIP/5555-00000027", "Macro Depth is 1") in new stack
    -- Executing [s@macro-user-callerid:23] GotoIf("SIP/5555-00000027", "1?report2:macroerror") in new stack
    -- Goto (macro-user-callerid,s,24)
    -- Executing [s@macro-user-callerid:24] GotoIf("SIP/5555-00000027", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:25] ExecIf("SIP/5555-00000027", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
    -- Executing [s@macro-user-callerid:26] Set("SIP/5555-00000027", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:27] GotoIf("SIP/5555-00000027", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,43)
    -- Executing [s@macro-user-callerid:43] Set("SIP/5555-00000027", "CALLERID(number)=5555") in new stack
    -- Executing [s@macro-user-callerid:44] Set("SIP/5555-00000027", "CALLERID(name)=5555") in new stack
    -- Executing [s@macro-user-callerid:45] GotoIf("SIP/5555-00000027", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:46] Set("SIP/5555-00000027", "CDR(cnam)=5555") in new stack
    -- Executing [s@macro-user-callerid:47] Set("SIP/5555-00000027", "CDR(cnum)=5555") in new stack
    -- Executing [s@macro-user-callerid:48] Set("SIP/5555-00000027", "CHANNEL(language)=en") in new stack
    -- Executing [*65@from-internal:7] GotoIf("SIP/5555-00000027", "1?app-speakextennum,en,1:app-speakextennum,en,1") in new stack
    -- Goto (app-speakextennum,en,1)
    -- Executing [en@app-speakextennum:1] Playback("SIP/5555-00000027", "your") in new stack
    -- <SIP/5555-00000027> Playing 'your.ulaw' (language 'en')
Retransmitting #4 (NAT) to [My Client IP]:52566:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Executing [en@app-speakextennum:2] Playback("SIP/5555-00000027", "extension") in new stack
    -- <SIP/5555-00000027> Playing 'extension.ulaw' (language 'en')
Really destroying SIP dialog 'vjmd4jeo7BIjdBqOtwYtuw..' Method: REGISTER
    -- Executing [en@app-speakextennum:3] Playback("SIP/5555-00000027", "number") in new stack
    -- <SIP/5555-00000027> Playing 'number.ulaw' (language 'en')
Retransmitting #5 (NAT) to [My Client IP]:52566:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Executing [en@app-speakextennum:4] Playback("SIP/5555-00000027", "is") in new stack
    -- <SIP/5555-00000027> Playing 'is.ulaw' (language 'en')
    -- Executing [en@app-speakextennum:5] SayDigits("SIP/5555-00000027", "5555") in new stack
    -- <SIP/5555-00000027> Playing 'digits/5.ulaw' (language 'en')
    -- <SIP/5555-00000027> Playing 'digits/5.ulaw' (language 'en')
    -- <SIP/5555-00000027> Playing 'digits/5.ulaw' (language 'en')
    -- <SIP/5555-00000027> Playing 'digits/5.ulaw' (language 'en')
Retransmitting #6 (NAT) to [My Client IP]:52566:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
[2020-05-12 20:36:03] WARNING[21959]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission UBKs1nTIc_v_ckIarWFcqw.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 7617ms with no response
[2020-05-12 20:36:03] WARNING[21959]: chan_sip.c:4166 retrans_pkt: Hanging up call UBKs1nTIc_v_ckIarWFcqw.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Spawn extension (app-speakextennum, en, 5) exited non-zero on 'SIP/5555-00000027'
Scheduling destruction of SIP dialog 'UBKs1nTIc_v_ckIarWFcqw..' in 7616 ms (Method: INVITE)
Reliably Transmitting (NAT) to [My Client IP]:52566:
BYE sip:[email protected]:36571 SIP/2.0
Via: SIP/2.0/UDP [My Server IP]:6116;branch=z9hG4bK3e40705e;rport
Max-Forwards: 70
From: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
To: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 102 BYE
User-Agent: FPBX-15.0.16.52(16.10.0)
Proxy-Authorization: Digest username="5555", realm="asterisk", algorithm=MD5, uri="sip:[FQDN of my Server ]", nonce="5dbbdc85", response="fdb70a489f709d119201a63c4d7530ee"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---

<--- SIP read from UDP:[My Client IP]:52566 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [My Server IP]:6116;branch=z9hG4bK3e40705e;rport=6116;received=[Second IP of my Server ]
Contact: <sip:[email protected]:36571>
To: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
From: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 102 BYE
User-Agent: SessionTalk 6.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'UBKs1nTIc_v_ckIarWFcqw..' Method: INVITE

No sound at all.
as we are behind an infrasctruture we are note managing , please note that
The SIP port is 6116 ( instead of the standard 5060)
I restricted for test for now the RTP ports range to 16600 to 16605 ( so enough to have 2 calls at the same time for now, I will ask later to a bigger range when I will get my FreePBX working )

Any insights about these lines ? it relates to RTP ports to be open for ex ? any method to troubleshoot ?
[2020-05-12 20:36:03] WARNING[21959]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission UBKs1nTIc_v_ckIarWFcqw… for seqno 2 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 7617ms with no response
[2020-05-12 20:36:03] WARNING[21959]: chan_sip.c:4166 retrans_pkt: Hanging up call UBKs1nTIc_v_ckIarWFcqw… - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Spawn extension (app-speakextennum, en, 5) exited non-zero on ‘SIP/5555-00000027’

Thanks for your insights

cynjut · May 12, 2020, 7:47pm

https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
would be a good place to start.

This is a classic NAT problem. Your NAT traffic is not making back to your phone.

This tells us that one of the NAT settings in the circuit is not working.

Registration happen on the Control Path (6116 in your example) and the RTP happens in two data streams between port 10000 and 20000. Clamping down the range doesn’t help in debugging. It could actually introduce a whole new set of problems.

Tell us a little more about your network - you don’t need to post addresses or passwords, just give is a feel for the topology. Where are the Firewalls, the PBX, the remote routers, the phones. Basically, anything that needs to be there for the call flow to work.

Also, be aware that you need to make sure that ALL of the NAT settings are correct. The ones in the extension config, the ones in the soft phones, the ones for the PBX itself.

lmon · May 12, 2020, 8:13pm

Hi,
the VOIP server is hosted in AWS.
There is a network load balancer which do the following port mapping :

Client ->> PUBLIC port 6116 —AWS NLB – > PBX internal IP / port 6116
Client ->> Public port 16001 --AWS NLB—> PBX Internal IP / port 16001
Client ->> Public port 16002 —AWS NLB—> PBX Internal IP / port 16002
Client ->> Public port 16XXX —AWS NLB—> PBX Internal IP / port XXXX

On the client side, I just tried using a sip client on my phone using 3G, so no issues on port blocking / nat …Etc ( just tested the sip client with another Freepbx install and it works wihtout problem )

cynjut · May 12, 2020, 8:19pm

I’m a ‘bare metal’ guy, so I’m at a disadvantage, but I know, for a fact, that almost everything you are doing is going to have to have NAT enabled. If the addresses change for any of the devices (non-routable to routable) you are going to have to make sure the NAT settings are correct. The logs you provided showed that the phones are probably not set up completely correctly, although they register (which is port 6116) you are not getting audio. This setup means that you need to make sure the PBX is also only using port 16001 and 16002. As a first step, I’d open up the range from 16000 to 17000, and make sure the numbers are both even. This allows both directions to work.

Even with 1:1 NAT, you still need to set the NAT rules up everywhere.

lmon · May 12, 2020, 8:23pm

OK, I will investigate with the LB admin.
My previous install was with a simple NAT.
Now , there is a LB…
Will keep the track posted,
Thanks a lot for your help and suggestions

dicko · May 12, 2020, 8:40pm

Just a note, RDP ports will always be an even number for the originating channel and the following odd port for channel B, all completed calls in Asterisk are two bridged calls so each "phone call’ needs 4 ports.

lmon · May 12, 2020, 9:12pm

@dicko , Hi , Thanks a lot.
Yes, that’s why I kept 5 ports for the RTP.
Will post the updates, I’m almost sure it is an AWS issue as the AWS LB doesn’t provides so much config options …)

dicko · May 12, 2020, 9:29pm

From acli

rtp set debug on

lmon · May 24, 2020, 10:01pm

Hi all,
Issue was with the AWS LB / NAT.
as almost everything is “automated” and we don’t have access to parameters, we are unable to define ( and even know ) what’s going on.
We have set direct port mapping, and it worked straight forward…
Thx to all for your inputs and help

system · June 24, 2020, 10:01pm

This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.