SIP - Phone registered - but no sounds

freepbx
configuration
asterisk
Tags: #<Tag:0x00007f701eb79388> #<Tag:0x00007f701eb79248> #<Tag:0x00007f701eb79040>

(Lmon) #1

Hi,
I already posted with some issues on the install. Now, everything is resolved and FreePBX runs ok.
My configuration is SIP only.
I created 2 extentions for now 5555 and 6666.
I share with you all the stuff I’m seeing to give as much info as possible :
1 - when I call from phone with 5555 to 6666, the call works but there is no sound
2 - when I call from phone 6666 to 5555 , the call do not reach
3 when I try on both phones *65 , echo test : no sounds

I configured in advaved sip : NAT_SIP to yes and callreinvite to “no”.

Here are the log from the console with SIP debug enabled ( I call, then hang up )
Any suggestions ?
Thanks !

ip-172-31-44-6CLI>
ip-172-31-44-6
CLI>
ip-172-31-44-6CLI>
ip-172-31-44-6
CLI>
ip-172-31-44-6CLI>
ip-172-31-44-6
CLI>
Retransmitting #1 (NAT) to [my_phone1_public_ip]:60797:
OPTIONS sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP SIP/2.0
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK24cb4e9c;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@my_pbx_public_ip:6116;tag=as787828ea
To: sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP
Contact: sip:Unknown@my_pbx_public_ip:6116
Call-ID: 1f67a6a40088d4c55147e19f459d5ad2@my_pbx_public_ip:6116
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.13.4(13.18.3)
Date: Wed, 26 Jun 2019 12:50:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


Retransmitting #2 (NAT) to [my_phone1_public_ip]:60797:
OPTIONS sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP SIP/2.0
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK24cb4e9c;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@my_pbx_public_ip:6116;tag=as787828ea
To: sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP
Contact: sip:Unknown@my_pbx_public_ip:6116
Call-ID: 1f67a6a40088d4c55147e19f459d5ad2@my_pbx_public_ip:6116
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.13.4(13.18.3)
Date: Wed, 26 Jun 2019 12:50:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


Retransmitting #3 (NAT) to [my_phone1_public_ip]:60797:
OPTIONS sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP SIP/2.0
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK24cb4e9c;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@my_pbx_public_ip:6116;tag=as787828ea
To: sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP
Contact: sip:Unknown@my_pbx_public_ip:6116
Call-ID: 1f67a6a40088d4c55147e19f459d5ad2@my_pbx_public_ip:6116
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.13.4(13.18.3)
Date: Wed, 26 Jun 2019 12:50:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:[my_phone1_public_ip]:60797 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK24cb4e9c;rport=6116
To: sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP
From: “Unknown” sip:Unknown@my_pbx_public_ip:6116;tag=as787828ea
Call-ID: 1f67a6a40088d4c55147e19f459d5ad2@my_pbx_public_ip:6116
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:[my_phone1_public_ip]:60797 —>
INVITE sip:6666@my_pbx_public_ip:6116 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—c442383f992d3513;rport
Max-Forwards: 70
Contact: sip:5555@192.168.1.65:60797;transport=UDP
To: sip:6666@my_pbx_public_ip:6116
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
Call-ID: qBwB_6hzd2q8868d-q-ubA…
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: 396

v=0
o=- 0 1 IN IP4 192.168.0.250
s=-
c=IN IP4 192.168.1.65
t=0 0
m=audio 4012 RTP/AVP 9 0 8 3 102 120 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
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_phone1_public_ip]:60797 (NAT)
Sending to [my_phone1_public_ip]:60797 (NAT)
Using INVITE request as basis request - qBwB_6hzd2q8868d-q-ubA…
Found peer ‘5555’ for ‘5555’ from [my_phone1_public_ip]:60797

<— Reliably Transmitting (NAT) to [my_phone1_public_ip]:60797 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—c442383f992d3513;received=[my_phone1_public_ip];rport=60797
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
To: sip:6666@my_pbx_public_ip:6116;tag=as123f2c1e
Call-ID: qBwB_6hzd2q8868d-q-ubA…
CSeq: 1 INVITE
Server: FPBX-14.0.13.4(13.18.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“429654d6”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘qBwB_6hzd2q8868d-q-ubA…’ in 6400 ms (Method: INVITE)
Retransmitting #1 (NAT) to [my_phone1_public_ip]:60797:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—c442383f992d3513;received=[my_phone1_public_ip];rport=60797
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
To: sip:6666@my_pbx_public_ip:6116;tag=as123f2c1e
Call-ID: qBwB_6hzd2q8868d-q-ubA…
CSeq: 1 INVITE
Server: FPBX-14.0.13.4(13.18.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“429654d6”
Content-Length: 0


<— SIP read from UDP:[my_phone1_public_ip]:60797 —>
ACK sip:6666@my_pbx_public_ip:6116 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—c442383f992d3513;rport
Max-Forwards: 70
To: sip:6666@my_pbx_public_ip:6116;tag=as123f2c1e
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
Call-ID: qBwB_6hzd2q8868d-q-ubA…
CSeq: 1 ACK
Content-Length: 0

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

<— SIP read from UDP:[my_phone1_public_ip]:60797 —>
INVITE sip:6666@my_pbx_public_ip:6116 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—5f529127dfa46e08;rport
Max-Forwards: 70
Contact: sip:5555@192.168.1.65:60797;transport=UDP
To: sip:6666@my_pbx_public_ip:6116
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
Call-ID: qBwB_6hzd2q8868d-q-ubA…
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=“429654d6”,uri=“sip:6666@my_pbx_public_ip:6116”,response=“0dd9fb65a4b22ad5d50fac0087417d94”,algorithm=MD5
Content-Length: 396

v=0
o=- 0 1 IN IP4 192.168.0.250
s=-
c=IN IP4 192.168.1.65
t=0 0
m=audio 4012 RTP/AVP 9 0 8 3 102 120 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
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_phone1_public_ip]:60797 (NAT)
Using INVITE request as basis request - qBwB_6hzd2q8868d-q-ubA…
Found peer ‘5555’ for ‘5555’ from [my_phone1_public_ip]:60797
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 102
Found RTP audio format 120
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 102
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|)
Peer audio RTP is at port 192.168.1.65:4012
Looking for 6666 in from-internal (domain my_pbx_public_ip)
sip_route_dump: route/path hop: sip:5555@192.168.1.65:60797;transport=UDP

<— Transmitting (NAT) to [my_phone1_public_ip]:60797 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—5f529127dfa46e08;received=[my_phone1_public_ip];rport=60797
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
To: sip:6666@my_pbx_public_ip:6116
Call-ID: qBwB_6hzd2q8868d-q-ubA…
CSeq: 2 INVITE
Server: FPBX-14.0.13.4(13.18.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:6666@my_pbx_public_ip:6116
Content-Length: 0

<------------>
Retransmitting #4 (NAT) to [my_phone1_public_ip]:60797:
OPTIONS sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP SIP/2.0
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK24cb4e9c;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@my_pbx_public_ip:6116;tag=as787828ea
To: sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP
Contact: sip:Unknown@my_pbx_public_ip:6116
Call-ID: 1f67a6a40088d4c55147e19f459d5ad2@my_pbx_public_ip:6116
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.13.4(13.18.3)
Date: Wed, 26 Jun 2019 12:50:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


Really destroying SIP dialog ‘1f67a6a40088d4c55147e19f459d5ad2@my_pbx_public_ip:6116’ Method: OPTIONS

<— SIP read from UDP:[my_phone1_public_ip]:60797 —>
ACK sip:6666@my_pbx_public_ip:6116 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—c442383f992d3513;rport
Max-Forwards: 70
To: sip:6666@my_pbx_public_ip:6116;tag=as123f2c1e
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
Call-ID: qBwB_6hzd2q8868d-q-ubA…
CSeq: 1 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
[2019-06-26 12:50:46] ERROR[2579][C-00027639]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2019-06-26 12:50:46] ERROR[2579][C-00027639]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2019-06-26 12:50:46] ERROR[2579][C-00027639]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2019-06-26 12:50:46] ERROR[2579][C-00027639]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2019-06-26 12:50:46] ERROR[2579][C-00027639]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2019-06-26 12:50:46] ERROR[2579][C-00027639]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2019-06-26 12:50:46] ERROR[2579][C-00027639]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2019-06-26 12:50:46] ERROR[2579][C-00027639]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2019-06-26 12:50:46] ERROR[2579][C-00027639]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2019-06-26 12:50:46] ERROR[2579][C-00027639]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2019-06-26 12:50:46] ERROR[2579][C-00027639]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2019-06-26 12:50:46] ERROR[2579][C-00027639]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
Audio is at 14732
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 92.184.97.97:51074:
INVITE sip:6666@10.238.179.18:51074;rinstance=3d98c5d903a67546;transport=UDP SIP/2.0
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK738c28f4;rport
Max-Forwards: 70
From: “5555” sip:5555@my_pbx_public_ip:6116;tag=as650f5db2
To: sip:6666@10.238.179.18:51074;rinstance=3d98c5d903a67546;transport=UDP
Contact: sip:5555@my_pbx_public_ip:6116
Call-ID: 1c3914dc73bc4b1c2f63f09d69ec3822@my_pbx_public_ip:6116
CSeq: 102 INVITE
User-Agent: FPBX-14.0.13.4(13.18.3)
Date: Wed, 26 Jun 2019 12:50:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: “5555” sip:5555@my_pbx_public_ip
Content-Type: application/sdp
Content-Length: 370

v=0
o=root 959313826 959313826 IN IP4 my_pbx_public_ip
s=Asterisk PBX 13.18.3~dfsg-1ubuntu4
c=IN IP4 my_pbx_public_ip
t=0 0
m=audio 14732 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


<— Transmitting (NAT) to [my_phone1_public_ip]:60797 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—5f529127dfa46e08;received=[my_phone1_public_ip];rport=60797
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
To: sip:6666@my_pbx_public_ip:6116;tag=as71565b16
Call-ID: qBwB_6hzd2q8868d-q-ubA…
CSeq: 2 INVITE
Server: FPBX-14.0.13.4(13.18.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:6666@my_pbx_public_ip:6116
P-Asserted-Identity: “6666” sip:6666@my_pbx_public_ip
Content-Length: 0

<------------>

<— SIP read from UDP:[my_phone1_public_ip]:60797 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK24cb4e9c;rport=6116
To: sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP
From: “Unknown” sip:Unknown@my_pbx_public_ip:6116;tag=as787828ea
Call-ID: 1f67a6a40088d4c55147e19f459d5ad2@my_pbx_public_ip:6116
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Retransmitting #1 (NAT) to 92.184.97.97:51074:
INVITE sip:6666@10.238.179.18:51074;rinstance=3d98c5d903a67546;transport=UDP SIP/2.0
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK738c28f4;rport
Max-Forwards: 70
From: “5555” sip:5555@my_pbx_public_ip:6116;tag=as650f5db2
To: sip:6666@10.238.179.18:51074;rinstance=3d98c5d903a67546;transport=UDP
Contact: sip:5555@my_pbx_public_ip:6116
Call-ID: 1c3914dc73bc4b1c2f63f09d69ec3822@my_pbx_public_ip:6116
CSeq: 102 INVITE
User-Agent: FPBX-14.0.13.4(13.18.3)
Date: Wed, 26 Jun 2019 12:50:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: “5555” sip:5555@my_pbx_public_ip
Content-Type: application/sdp
Content-Length: 370

v=0
o=root 959313826 959313826 IN IP4 my_pbx_public_ip
s=Asterisk PBX 13.18.3~dfsg-1ubuntu4
c=IN IP4 my_pbx_public_ip
t=0 0
m=audio 14732 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
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:92.184.97.97:51074 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK738c28f4;rport=6116
To: sip:6666@10.238.179.18:51074;rinstance=3d98c5d903a67546;transport=UDP
From: “5555” sip:5555@my_pbx_public_ip:6116;tag=as650f5db2
Call-ID: 1c3914dc73bc4b1c2f63f09d69ec3822@my_pbx_public_ip:6116
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:92.184.97.97:51074 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK738c28f4;rport=6116
To: sip:6666@10.238.179.18:51074;rinstance=3d98c5d903a67546;transport=UDP
From: “5555” sip:5555@my_pbx_public_ip:6116;tag=as650f5db2
Call-ID: 1c3914dc73bc4b1c2f63f09d69ec3822@my_pbx_public_ip:6116
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:92.184.97.97:51074 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK738c28f4;rport=6116
Contact: sip:6666@10.238.179.18:51074
To: sip:6666@10.238.179.18:51074;rinstance=3d98c5d903a67546;transport=UDP;tag=17ec0701
From: “5555” sip:5555@my_pbx_public_ip:6116;tag=as650f5db2
Call-ID: 1c3914dc73bc4b1c2f63f09d69ec3822@my_pbx_public_ip:6116
CSeq: 102 INVITE
User-Agent: SessionTalk 6.0
Content-Length: 0

<------------->
— (9 headers 0 lines) —
sip_route_dump: route/path hop: sip:6666@10.238.179.18:51074

<— SIP read from UDP:92.184.97.97:51074 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK738c28f4;rport=6116
Contact: sip:6666@10.238.179.18:51074
To: sip:6666@10.238.179.18:51074;rinstance=3d98c5d903a67546;transport=UDP;tag=17ec0701
From: “5555” sip:5555@my_pbx_public_ip:6116;tag=as650f5db2
Call-ID: 1c3914dc73bc4b1c2f63f09d69ec3822@my_pbx_public_ip:6116
CSeq: 102 INVITE
User-Agent: SessionTalk 6.0
Content-Length: 0

<------------->
— (9 headers 0 lines) —
sip_route_dump: route/path hop: sip:6666@10.238.179.18:51074

<— Transmitting (NAT) to [my_phone1_public_ip]:60797 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—5f529127dfa46e08;received=[my_phone1_public_ip];rport=60797
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
To: sip:6666@my_pbx_public_ip:6116;tag=as71565b16
Call-ID: qBwB_6hzd2q8868d-q-ubA…
CSeq: 2 INVITE
Server: FPBX-14.0.13.4(13.18.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:6666@my_pbx_public_ip:6116
Content-Length: 0

<------------>

<— SIP read from UDP:92.184.97.97:51074 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK738c28f4;rport=6116
Require: timer
Contact: sip:6666@10.238.179.18:51074
To: sip:6666@10.238.179.18:51074;rinstance=3d98c5d903a67546;transport=UDP;tag=17ec0701
From: “5555” sip:5555@my_pbx_public_ip:6116;tag=as650f5db2
Call-ID: 1c3914dc73bc4b1c2f63f09d69ec3822@my_pbx_public_ip:6116
CSeq: 102 INVITE
Session-Expires: 1800;refresher=uac
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: 185

v=0
o=- 0 1 IN IP4 192.168.0.250
s=-
c=IN IP4 10.238.179.18
t=0 0
m=audio 4004 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (15 headers 10 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.238.179.18:4004
sip_route_dump: route/path hop: sip:6666@10.238.179.18:51074
Transmitting (NAT) to 92.184.97.97:51074:
ACK sip:6666@10.238.179.18:51074 SIP/2.0
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK1582aee1;rport
Max-Forwards: 70
From: “5555” sip:5555@my_pbx_public_ip:6116;tag=as650f5db2
To: sip:6666@10.238.179.18:51074;rinstance=3d98c5d903a67546;transport=UDP;tag=17ec0701
Contact: sip:5555@my_pbx_public_ip:6116
Call-ID: 1c3914dc73bc4b1c2f63f09d69ec3822@my_pbx_public_ip:6116
CSeq: 102 ACK
User-Agent: FPBX-14.0.13.4(13.18.3)
Content-Length: 0


Audio is at 10054
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_phone1_public_ip]:60797 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—5f529127dfa46e08;received=[my_phone1_public_ip];rport=60797
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
To: sip:6666@my_pbx_public_ip:6116;tag=as71565b16
Call-ID: qBwB_6hzd2q8868d-q-ubA…
CSeq: 2 INVITE
Server: FPBX-14.0.13.4(13.18.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:6666@my_pbx_public_ip:6116
P-Asserted-Identity: “6666” sip:6666@my_pbx_public_ip
Content-Type: application/sdp
Require: timer
Content-Length: 341

v=0
o=root 1833047560 1833047560 IN IP4 my_pbx_public_ip
s=Asterisk PBX 13.18.3~dfsg-1ubuntu4
c=IN IP4 my_pbx_public_ip
t=0 0
m=audio 10054 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_phone1_public_ip]:60797:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—5f529127dfa46e08;received=[my_phone1_public_ip];rport=60797
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
To: sip:6666@my_pbx_public_ip:6116;tag=as71565b16
Call-ID: qBwB_6hzd2q8868d-q-ubA…
CSeq: 2 INVITE
Server: FPBX-14.0.13.4(13.18.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:6666@my_pbx_public_ip:6116
P-Asserted-Identity: “6666” sip:6666@my_pbx_public_ip
Content-Type: application/sdp
Require: timer
Content-Length: 341

v=0
o=root 1833047560 1833047560 IN IP4 my_pbx_public_ip
s=Asterisk PBX 13.18.3~dfsg-1ubuntu4
c=IN IP4 my_pbx_public_ip
t=0 0
m=audio 10054 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_phone1_public_ip]:60797 —>
ACK sip:6666@my_pbx_public_ip:6116 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—55e98467d1d23c71;rport
Max-Forwards: 70
To: sip:6666@my_pbx_public_ip:6116;tag=as71565b16
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
Call-ID: qBwB_6hzd2q8868d-q-ubA…
CSeq: 2 ACK
User-Agent: SessionTalk 6.0
Content-Length: 0

<------------->
— (9 headers 0 lines) —
[2019-06-26 12:50:53] WARNING[2579][C-00027639]: res_rtp_asterisk.c:5272 ast_rtp_read: RTP Read too short

<— SIP read from UDP:[my_phone1_public_ip]:60797 —>
ACK sip:6666@my_pbx_public_ip:6116 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—55e98467d1d23c71;rport
Max-Forwards: 70
To: sip:6666@my_pbx_public_ip:6116;tag=as71565b16
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
Call-ID: qBwB_6hzd2q8868d-q-ubA…
CSeq: 2 ACK
User-Agent: SessionTalk 6.0
Content-Length: 0

<------------->
— (9 headers 0 lines) —
[2019-06-26 12:50:53] WARNING[2582][C-0002763a]: func_channel.c:465 func_channel_read: Unknown or unavailable item requested: ‘recvip’
[2019-06-26 12:50:53] WARNING[2582][C-0002763a]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from "
Reliably Transmitting (NAT) to [my_phone1_public_ip]:60797:
OPTIONS sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP SIP/2.0
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK10c8a431;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@my_pbx_public_ip:6116;tag=as57ab3771
To: sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP
Contact: sip:Unknown@my_pbx_public_ip:6116
Call-ID: 4e50c31b787be15d605842805fa33352@my_pbx_public_ip:6116
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.13.4(13.18.3)
Date: Wed, 26 Jun 2019 12:50:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


Retransmitting #1 (NAT) to [my_phone1_public_ip]:60797:
OPTIONS sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP SIP/2.0
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK10c8a431;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@my_pbx_public_ip:6116;tag=as57ab3771
To: sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP
Contact: sip:Unknown@my_pbx_public_ip:6116
Call-ID: 4e50c31b787be15d605842805fa33352@my_pbx_public_ip:6116
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.13.4(13.18.3)
Date: Wed, 26 Jun 2019 12:50:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


Retransmitting #2 (NAT) to [my_phone1_public_ip]:60797:
OPTIONS sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP SIP/2.0
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK10c8a431;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@my_pbx_public_ip:6116;tag=as57ab3771
To: sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP
Contact: sip:Unknown@my_pbx_public_ip:6116
Call-ID: 4e50c31b787be15d605842805fa33352@my_pbx_public_ip:6116
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.13.4(13.18.3)
Date: Wed, 26 Jun 2019 12:50:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


Retransmitting #3 (NAT) to [my_phone1_public_ip]:60797:
OPTIONS sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP SIP/2.0
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK10c8a431;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@my_pbx_public_ip:6116;tag=as57ab3771
To: sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP
Contact: sip:Unknown@my_pbx_public_ip:6116
Call-ID: 4e50c31b787be15d605842805fa33352@my_pbx_public_ip:6116
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.13.4(13.18.3)
Date: Wed, 26 Jun 2019 12:50:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:[my_phone1_public_ip]:60797 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK10c8a431;rport=6116
To: sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP
From: “Unknown” sip:Unknown@my_pbx_public_ip:6116;tag=as57ab3771
Call-ID: 4e50c31b787be15d605842805fa33352@my_pbx_public_ip:6116
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Retransmitting #4 (NAT) to [my_phone1_public_ip]:60797:
OPTIONS sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP SIP/2.0
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK10c8a431;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@my_pbx_public_ip:6116;tag=as57ab3771
To: sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP
Contact: sip:Unknown@my_pbx_public_ip:6116
Call-ID: 4e50c31b787be15d605842805fa33352@my_pbx_public_ip:6116
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.13.4(13.18.3)
Date: Wed, 26 Jun 2019 12:50:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


Really destroying SIP dialog ‘4e50c31b787be15d605842805fa33352@my_pbx_public_ip:6116’ Method: OPTIONS

<— SIP read from UDP:[my_phone1_public_ip]:60797 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK10c8a431;rport=6116
To: sip:5555@192.168.1.65:60797;rinstance=cd2270929f543584;transport=UDP
From: “Unknown” sip:Unknown@my_pbx_public_ip:6116;tag=as57ab3771
Call-ID: 4e50c31b787be15d605842805fa33352@my_pbx_public_ip:6116
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:[my_phone1_public_ip]:60797 —>

<------------->

<— SIP read from UDP:[my_phone1_public_ip]:60797 —>
BYE sip:6666@my_pbx_public_ip:6116 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—309bf12a735dd860;rport
Max-Forwards: 70
Contact: sip:5555@192.168.1.65:60797;transport=UDP
To: sip:6666@my_pbx_public_ip:6116;tag=as71565b16
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
Call-ID: qBwB_6hzd2q8868d-q-ubA…
CSeq: 3 BYE
User-Agent: SessionTalk 6.0
Authorization: Digest username=“5555”,realm=“asterisk”,nonce=“429654d6”,uri=“sip:6666@my_pbx_public_ip:6116”,response=“4926cf6557774d5708665642d8ed506f”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to [my_phone1_public_ip]:60797 (NAT)
Scheduling destruction of SIP dialog ‘qBwB_6hzd2q8868d-q-ubA…’ in 6400 ms (Method: BYE)

<— Transmitting (NAT) to [my_phone1_public_ip]:60797 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.65:60797;branch=z9hG4bK-524287-1—309bf12a735dd860;received=[my_phone1_public_ip];rport=60797
From: "5555"sip:5555@my_pbx_public_ip:6116;tag=30c45870
To: sip:6666@my_pbx_public_ip:6116;tag=as71565b16
Call-ID: qBwB_6hzd2q8868d-q-ubA…
CSeq: 3 BYE
Server: FPBX-14.0.13.4(13.18.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1c3914dc73bc4b1c2f63f09d69ec3822@my_pbx_public_ip:6116’ in 21184 ms (Method: INVITE)
Reliably Transmitting (NAT) to 92.184.97.97:51074:
BYE sip:6666@10.238.179.18:51074 SIP/2.0
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK62872669;rport
Max-Forwards: 70
From: “5555” sip:5555@my_pbx_public_ip:6116;tag=as650f5db2
To: sip:6666@10.238.179.18:51074;rinstance=3d98c5d903a67546;transport=UDP;tag=17ec0701
Call-ID: 1c3914dc73bc4b1c2f63f09d69ec3822@my_pbx_public_ip:6116
CSeq: 103 BYE
User-Agent: FPBX-14.0.13.4(13.18.3)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<— SIP read from UDP:92.184.97.97:51074 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP my_pbx_public_ip:6116;branch=z9hG4bK62872669;rport=6116
Contact: sip:6666@10.238.179.18:51074
To: sip:6666@10.238.179.18:51074;rinstance=3d98c5d903a67546;transport=UDP;tag=17ec0701
From: “5555” sip:5555@my_pbx_public_ip:6116;tag=as650f5db2
Call-ID: 1c3914dc73bc4b1c2f63f09d69ec3822@my_pbx_public_ip:6116
CSeq: 103 BYE
User-Agent: SessionTalk 6.0
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘1c3914dc73bc4b1c2f63f09d69ec3822@my_pbx_public_ip:6116’ Method: INVITE
[2019-06-26 12:51:03] WARNING[31543]: res_pjsip_registrar.c:963 registrar_on_rx_request: Endpoint ‘anonymous’ has no configured AORs
[2019-06-26 12:51:04] WARNING[2644][C-0002763b]: func_channel.c:465 func_channel_read: Unknown or unavailable item requested: ‘recvip’
[2019-06-26 12:51:04] WARNING[2644][C-0002763b]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from "
ip-172-31-44-6CLI> sip set debug off
SIP Debugging Disabled
[2019-06-26 12:51:17] WARNING[2645][C-0002763c]: func_channel.c:465 func_channel_read: Unknown or unavailable item requested: ‘recvip’
[2019-06-26 12:51:17] WARNING[2645][C-0002763c]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from "
ip-172-31-44-6
CLI>


(Dave Burgess) #2

This looks a lot like your system isn’t configured to recognize the local network.


(Lmon) #3

I discovered that I have many external connection unwanted. I activated fail2ban, I think this is a “parasite message” no ? as we can establish the registration and make the call ?


(Dave Burgess) #4

You should also make sure that Allow Anonymous is turned off in the Advanced settings.

Tell us more about your server. Local or cloud? Version? Integrated Firewall installed and set up?

One-way audio, even when it’s both ways, is usually a problem with the RTP ports (10000-20000) and NAT. Tell us more about your NAT setup and how your network is configured.


(Lmon) #5

Allow anonymous is turned of yes
Server is hosted in AWS
Firewall is “open”, except for the fail2ban IP which are detected.
Found the issue … I need to be more restrictive on the ports to open in AWS :
if I define 16384-32767 in UDP, it works … I defined the range to wide and it was not applied …
Thanks !


(Dave Burgess) #6

You need to either configure your PBX to use those addresses, or use 10000-20000 UDP. Also, you may need to redirect those ports to your PBX server for access from your ITSP.


(system) closed #7

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.