Retransmission timeout reached on transmission outbound

#1

Hi Guys,

Need your help with an odd problem.
When we do an inbound call everything works perfectly.

An outbound call doesn’t seem to work.
We keep getting the following error.

Retransmission timeout reached on transmission a9s9gsc-qPxN00amC-J8tw.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response

After this the call is hanged up.

We have read all the information on the website, that is provided with the error, but this doesn’t seem to fix the problem.
Does anybody have any ideas about what we are doing wrong here?

Help would be highly appreciated.

Our outbound Settings.

[FUSION_INBOUND](!)
type        = friend
context     = context_tok
insecure    = port,invite
canreinvite = no
disallow    = all
allow       = ulaw
dtmfmode    = inband
qualify     = yes
allowguest  = yes

[FUSION_OUTBOUND](!)
type        = friend
context     = default
insecure    = port,invite
canreinvite = no
disallow    = all
allow       = ulaw
dtmfmode    = inband
;nat         = force_rport,comedia
nat         = never
qualify     = yes
allowguest  = yes

The extension log

exten => _0.,1,noop(outbound)
same => n,dial(SIP/outbound_1_1/${EXTEN})
same => n,hangup

The current verbose log


 
 
 v=0
 o=Z 148814912 0 IN IP4 192.168.100.234
 s=Z
 c=IN IP4 192.168.100.234
 t=0 0
 m=audio 8000 RTP/AVP 106 9 3 111 0 8 97 110 112 98 101 100 99 102
 a=rtpmap:106 opus/48000/2
 a=fmtp:106 minptime=20; cbr=1; maxaveragebitrate=40000; useinbandfec=1
 a=rtpmap:111 speex/16000
 a=rtpmap:97 iLBC/8000
 a=fmtp:97 mode=20
 a=rtpmap:110 speex/8000
 a=rtpmap:112 speex/32000
 a=rtpmap:98 telephone-event/48000
 a=fmtp:98 0-16
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=rtpmap:100 telephone-event/16000
 a=fmtp:100 0-16
 a=rtpmap:99 telephone-event/32000
 a=fmtp:99 0-16
 a=rtpmap:102 G726-32/8000
 a=sendrecv
 <------------->
 --- (14 headers 23 lines) ---
 Ignoring this INVITE request
 
 <--- Transmitting (NAT) to 111.108.30.208:61654 --->
 SIP/2.0 100 Trying
 Via: SIP/2.0/UDP 111.108.30.208:60463;branch=z9hG4bK-524287-1---637b9b483d34a6d0;received=111.108.30.208;rport=61654
 From: <sip:1_1_1_1@52.194.253.25:15060;transport=UDP>;tag=a93b3860
 To: <sip:05052424142@52.194.253.25:15060;transport=UDP>
 Call-ID: a9s9gsc-qPxN00amC-J8tw..
 CSeq: 2 INVITE
 Server: Asterisk PBX 13.11.0-rc1
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
 Supported: replaces, timer
 Contact: <sip:05052424142@52.194.253.25:15060>
 Content-Length: 0
 
 
 <------------>
 
 <--- SIP read from UDP:61.213.230.153:5060 --->
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK67862633;received=52.194.253.25;rport=15060
 From: <sip:52431824@smart.0038.net:15060>;tag=as1d7e81f3
 To: <sip:05052424142@smart.0038.net>;tag=as36f57cd5
 Call-ID: 1edcbdc676fb9def22d9ddcf04e593b9@smart.0038.net
 CSeq: 103 INVITE
 Server: Fusion Open MGW 1.0
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
 Supported: replaces, timer
 Session-Expires: 1800;refresher=uas
 Contact: <sip:05052424142@61.213.230.153:5060>
 Content-Type: application/sdp
 Require: timer
 Content-Length: 161
 
 v=0
 o=root 1146834922 1146834922 IN IP4 61.213.230.9
 s=-
 c=IN IP4 61.213.230.9
 t=0 0
 m=audio 13102 RTP/AVP 0
 a=rtpmap:0 PCMU/8000
 a=ptime:20
 a=sendrecv
 <------------->
 --- (14 headers 9 lines) ---
 Found RTP audio format 0
 Found audio description format PCMU for ID 0
 Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
 Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
 Peer audio RTP is at port 61.213.230.9:13102
 sip_route_dump: route/path hop: <sip:05052424142@61.213.230.153:5060>
 set_destination: Parsing <sip:05052424142@61.213.230.153:5060> for address/port to send to
 set_destination: set destination to 61.213.230.153:5060
 Transmitting (no NAT) to 61.213.230.153:5060:
 ACK sip:05052424142@61.213.230.153:5060 SIP/2.0
 Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK4fec0890
 Max-Forwards: 70
 From: <sip:52431824@smart.0038.net:15060>;tag=as1d7e81f3
 To: <sip:05052424142@smart.0038.net>;tag=as36f57cd5
 Contact: <sip:52431824@52.194.253.25:15060>
 Call-ID: 1edcbdc676fb9def22d9ddcf04e593b9@smart.0038.net
 CSeq: 103 ACK
 User-Agent: Asterisk PBX 13.11.0-rc1
 Content-Length: 0
 
 
 ---
     -- SIP/outbound_1_1-00000001 answered SIP/1_1_1_1-00000000
 Audio is at 20494
 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 111.108.30.208:61654 --->
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP 111.108.30.208:60463;branch=z9hG4bK-524287-1---637b9b483d34a6d0;received=111.108.30.208;rport=61654
 From: <sip:1_1_1_1@52.194.253.25:15060;transport=UDP>;tag=a93b3860
 To: <sip:05052424142@52.194.253.25:15060;transport=UDP>;tag=as514844ef
 Call-ID: a9s9gsc-qPxN00amC-J8tw..
 CSeq: 2 INVITE
 Server: Asterisk PBX 13.11.0-rc1
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
 Supported: replaces, timer
 Contact: <sip:05052424142@52.194.253.25:15060>
 Content-Type: application/sdp
 Content-Length: 305
 
 v=0
 o=root 1876765095 1876765095 IN IP4 52.194.253.25
 s=Asterisk PBX 13.11.0-rc1
 c=IN IP4 52.194.253.25
 t=0 0
 m=audio 20494 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=ptime:20
 a=maxptime:150
 a=sendrecv
 
 <------------>
     -- Channel SIP/outbound_1_1-00000001 joined 'simple_bridge' basic-bridge <39afb92d-e7da-411b-8251-eea995876826>
     -- Channel SIP/1_1_1_1-00000000 joined 'simple_bridge' basic-bridge <39afb92d-e7da-411b-8251-eea995876826>
        > 0x7fc7b400a490 -- Probation passed - setting RTP source address to 61.213.230.9:13102
 Retransmitting #1 (NAT) to 111.108.30.208:61654:
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP 111.108.30.208:60463;branch=z9hG4bK-524287-1---637b9b483d34a6d0;received=111.108.30.208;rport=61654
 From: <sip:1_1_1_1@52.194.253.25:15060;transport=UDP>;tag=a93b3860
 To: <sip:05052424142@52.194.253.25:15060;transport=UDP>;tag=as514844ef
 Call-ID: a9s9gsc-qPxN00amC-J8tw..
 CSeq: 2 INVITE
 Server: Asterisk PBX 13.11.0-rc1
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
 Supported: replaces, timer
 Contact: <sip:05052424142@52.194.253.25:15060>
 Content-Type: application/sdp
 Content-Length: 305
 
 v=0
 o=root 1876765095 1876765095 IN IP4 52.194.253.25
 s=Asterisk PBX 13.11.0-rc1
 c=IN IP4 52.194.253.25
 t=0 0
 m=audio 20494 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=ptime:20
 a=maxptime:150
 a=sendrecv
 
 ---
 Retransmitting #2 (NAT) to 111.108.30.208:61654:
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP 111.108.30.208:60463;branch=z9hG4bK-524287-1---637b9b483d34a6d0;received=111.108.30.208;rport=61654
 From: <sip:1_1_1_1@52.194.253.25:15060;transport=UDP>;tag=a93b3860
 To: <sip:05052424142@52.194.253.25:15060;transport=UDP>;tag=as514844ef
 Call-ID: a9s9gsc-qPxN00amC-J8tw..
 CSeq: 2 INVITE
 Server: Asterisk PBX 13.11.0-rc1
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
 Supported: replaces, timer
 Contact: <sip:05052424142@52.194.253.25:15060>
 Content-Type: application/sdp
 Content-Length: 305
 
 v=0
 o=root 1876765095 1876765095 IN IP4 52.194.253.25
 s=Asterisk PBX 13.11.0-rc1
 c=IN IP4 52.194.253.25
 t=0 0
 m=audio 20494 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=ptime:20
 a=maxptime:150
 a=sendrecv
 
 ---
 
 <--- SIP read from UDP:61.213.230.153:5060 --->
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK67862633;received=52.194.253.25;rport=15060
 From: <sip:52431824@smart.0038.net:15060>;tag=as1d7e81f3
 To: <sip:05052424142@smart.0038.net>;tag=as36f57cd5
 Call-ID: 1edcbdc676fb9def22d9ddcf04e593b9@smart.0038.net
 CSeq: 103 INVITE
 Server: Fusion Open MGW 1.0
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
 Supported: replaces, timer
 Session-Expires: 1800;refresher=uas
 Contact: <sip:05052424142@61.213.230.153:5060>
 Content-Type: application/sdp
 Require: timer
 Content-Length: 161
 
 v=0
 o=root 1146834922 1146834922 IN IP4 61.213.230.9
 s=-
 c=IN IP4 61.213.230.9
 t=0 0
 m=audio 13102 RTP/AVP 0
 a=rtpmap:0 PCMU/8000
 a=ptime:20
 a=sendrecv
 <------------->
 --- (14 headers 9 lines) ---
 set_destination: Parsing <sip:05052424142@61.213.230.153:5060> for address/port to send to
 set_destination: set destination to 61.213.230.153:5060
 Transmitting (no NAT) to 61.213.230.153:5060:
 ACK sip:05052424142@61.213.230.153:5060 SIP/2.0
 Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK4c9fb90b
 Max-Forwards: 70
 From: <sip:52431824@smart.0038.net:15060>;tag=as1d7e81f3
 To: <sip:05052424142@smart.0038.net>;tag=as36f57cd5
 Contact: <sip:52431824@52.194.253.25:15060>
 Call-ID: 1edcbdc676fb9def22d9ddcf04e593b9@smart.0038.net
 CSeq: 103 ACK
 User-Agent: Asterisk PBX 13.11.0-rc1
 Content-Length: 0
 
 
 ---
[Apr 12 14:47:24] Retransmitting #3 (NAT) to 111.108.30.208:61654:
[Apr 12 14:47:24] SIP/2.0 200 OK
[Apr 12 14:47:24] Via: SIP/2.0/UDP 111.108.30.208:60463;branch=z9hG4bK-524287-1---637b9b483d34a6d0;received=111.108.30.208;rport=61654
[Apr 12 14:47:24] From: <sip:1_1_1_1@52.194.253.25:15060;transport=UDP>;tag=a93b3860
[Apr 12 14:47:24] To: <sip:05052424142@52.194.253.25:15060;transport=UDP>;tag=as514844ef
[Apr 12 14:47:24] Call-ID: a9s9gsc-qPxN00amC-J8tw..
[Apr 12 14:47:24] CSeq: 2 INVITE
[Apr 12 14:47:24] Server: Asterisk PBX 13.11.0-rc1
[Apr 12 14:47:24] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Apr 12 14:47:24] Supported: replaces, timer
[Apr 12 14:47:24] Contact: <sip:05052424142@52.194.253.25:15060>
[Apr 12 14:47:24] Content-Type: application/sdp
[Apr 12 14:47:24] Content-Length: 305
[Apr 12 14:47:24] 
[Apr 12 14:47:24] v=0
[Apr 12 14:47:24] o=root 1876765095 1876765095 IN IP4 52.194.253.25
[Apr 12 14:47:24] s=Asterisk PBX 13.11.0-rc1
[Apr 12 14:47:24] c=IN IP4 52.194.253.25
[Apr 12 14:47:24] t=0 0
[Apr 12 14:47:24] m=audio 20494 RTP/AVP 0 8 3 101
[Apr 12 14:47:24] a=rtpmap:0 PCMU/8000
[Apr 12 14:47:24] a=rtpmap:8 PCMA/8000
[Apr 12 14:47:24] a=rtpmap:3 GSM/8000
[Apr 12 14:47:24] a=rtpmap:101 telephone-event/8000
[Apr 12 14:47:24] a=fmtp:101 0-16
[Apr 12 14:47:24] a=ptime:20
[Apr 12 14:47:24] a=maxptime:150
[Apr 12 14:47:24] a=sendrecv
[Apr 12 14:47:24] 
[Apr 12 14:47:24] ---
[Apr 12 14:47:24] Retransmitting #4 (NAT) to 111.108.30.208:61654:
[Apr 12 14:47:24] SIP/2.0 200 OK
[Apr 12 14:47:24] Via: SIP/2.0/UDP 111.108.30.208:60463;branch=z9hG4bK-524287-1---637b9b483d34a6d0;received=111.108.30.208;rport=61654
[Apr 12 14:47:24] From: <sip:1_1_1_1@52.194.253.25:15060;transport=UDP>;tag=a93b3860
[Apr 12 14:47:24] To: <sip:05052424142@52.194.253.25:15060;transport=UDP>;tag=as514844ef
[Apr 12 14:47:24] Call-ID: a9s9gsc-qPxN00amC-J8tw..
[Apr 12 14:47:24] CSeq: 2 INVITE
[Apr 12 14:47:24] Server: Asterisk PBX 13.11.0-rc1
[Apr 12 14:47:24] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Apr 12 14:47:24] Supported: replaces, timer
[Apr 12 14:47:24] Contact: <sip:05052424142@52.194.253.25:15060>
[Apr 12 14:47:24] Content-Type: application/sdp
[Apr 12 14:47:24] Content-Length: 305
[Apr 12 14:47:24] 
[Apr 12 14:47:24] v=0
[Apr 12 14:47:24] o=root 1876765095 1876765095 IN IP4 52.194.253.25
[Apr 12 14:47:24] s=Asterisk PBX 13.11.0-rc1
[Apr 12 14:47:24] c=IN IP4 52.194.253.25
[Apr 12 14:47:24] t=0 0
[Apr 12 14:47:24] m=audio 20494 RTP/AVP 0 8 3 101
[Apr 12 14:47:24] a=rtpmap:0 PCMU/8000
[Apr 12 14:47:24] a=rtpmap:8 PCMA/8000
[Apr 12 14:47:24] a=rtpmap:3 GSM/8000
[Apr 12 14:47:24] a=rtpmap:101 telephone-event/8000
[Apr 12 14:47:24] a=fmtp:101 0-16
[Apr 12 14:47:24] a=ptime:20
[Apr 12 14:47:24] a=maxptime:150
[Apr 12 14:47:24] a=sendrecv
[Apr 12 14:47:24] 
[Apr 12 14:47:24] ---
[Apr 12 14:47:26] Retransmitting #5 (NAT) to 111.108.30.208:61654:
[Apr 12 14:47:26] SIP/2.0 200 OK
[Apr 12 14:47:26] Via: SIP/2.0/UDP 111.108.30.208:60463;branch=z9hG4bK-524287-1---637b9b483d34a6d0;received=111.108.30.208;rport=61654
[Apr 12 14:47:26] From: <sip:1_1_1_1@52.194.253.25:15060;transport=UDP>;tag=a93b3860
[Apr 12 14:47:26] To: <sip:05052424142@52.194.253.25:15060;transport=UDP>;tag=as514844ef
[Apr 12 14:47:26] Call-ID: a9s9gsc-qPxN00amC-J8tw..
[Apr 12 14:47:26] CSeq: 2 INVITE
[Apr 12 14:47:26] Server: Asterisk PBX 13.11.0-rc1
[Apr 12 14:47:26] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Apr 12 14:47:26] Supported: replaces, timer
[Apr 12 14:47:26] Contact: <sip:05052424142@52.194.253.25:15060>
[Apr 12 14:47:26] Content-Type: application/sdp
[Apr 12 14:47:26] Content-Length: 305
[Apr 12 14:47:26] 
[Apr 12 14:47:26] v=0
[Apr 12 14:47:26] o=root 1876765095 1876765095 IN IP4 52.194.253.25
[Apr 12 14:47:26] s=Asterisk PBX 13.11.0-rc1
[Apr 12 14:47:26] c=IN IP4 52.194.253.25
[Apr 12 14:47:26] t=0 0
[Apr 12 14:47:26] m=audio 20494 RTP/AVP 0 8 3 101
[Apr 12 14:47:26] a=rtpmap:0 PCMU/8000
[Apr 12 14:47:26] a=rtpmap:8 PCMA/8000
[Apr 12 14:47:26] a=rtpmap:3 GSM/8000
[Apr 12 14:47:26] a=rtpmap:101 telephone-event/8000
[Apr 12 14:47:26] a=fmtp:101 0-16
[Apr 12 14:47:26] a=ptime:20
[Apr 12 14:47:26] a=maxptime:150
[Apr 12 14:47:26] a=sendrecv
[Apr 12 14:47:26] 
[Apr 12 14:47:26] ---
[Apr 12 14:47:28] 
[Apr 12 14:47:28] <--- SIP read from UDP:111.108.30.208:60463 --->
[Apr 12 14:47:28] 
[Apr 12 14:47:28] 
[Apr 12 14:47:28] <------------->
[Apr 12 14:47:29] Really destroying SIP dialog 'AcIpz9Mt93nNZ_slZ6wmHA..' Method: REGISTER
[Apr 12 14:47:29] Retransmitting #6 (NAT) to 111.108.30.208:61654:
[Apr 12 14:47:29] SIP/2.0 200 OK
[Apr 12 14:47:29] Via: SIP/2.0/UDP 111.108.30.208:60463;branch=z9hG4bK-524287-1---637b9b483d34a6d0;received=111.108.30.208;rport=61654
[Apr 12 14:47:29] From: <sip:1_1_1_1@52.194.253.25:15060;transport=UDP>;tag=a93b3860
[Apr 12 14:47:29] To: <sip:05052424142@52.194.253.25:15060;transport=UDP>;tag=as514844ef
[Apr 12 14:47:29] Call-ID: a9s9gsc-qPxN00amC-J8tw..
[Apr 12 14:47:29] CSeq: 2 INVITE
[Apr 12 14:47:29] Server: Asterisk PBX 13.11.0-rc1
[Apr 12 14:47:29] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Apr 12 14:47:29] Supported: replaces, timer
[Apr 12 14:47:29] Contact: <sip:05052424142@52.194.253.25:15060>
[Apr 12 14:47:29] Content-Type: application/sdp
[Apr 12 14:47:29] Content-Length: 305
[Apr 12 14:47:29] 
[Apr 12 14:47:29] v=0
[Apr 12 14:47:29] o=root 1876765095 1876765095 IN IP4 52.194.253.25
[Apr 12 14:47:29] s=Asterisk PBX 13.11.0-rc1
[Apr 12 14:47:29] c=IN IP4 52.194.253.25
[Apr 12 14:47:29] t=0 0
[Apr 12 14:47:29] m=audio 20494 RTP/AVP 0 8 3 101
[Apr 12 14:47:29] a=rtpmap:0 PCMU/8000
[Apr 12 14:47:29] a=rtpmap:8 PCMA/8000
[Apr 12 14:47:29] a=rtpmap:3 GSM/8000
[Apr 12 14:47:29] a=rtpmap:101 telephone-event/8000
[Apr 12 14:47:29] a=fmtp:101 0-16
[Apr 12 14:47:29] a=ptime:20
[Apr 12 14:47:29] a=maxptime:150
[Apr 12 14:47:29] a=sendrecv
[Apr 12 14:47:29] 
[Apr 12 14:47:29] ---
[Apr 12 14:47:29] WARNING[28896]: chan_sip.c:4059 retrans_pkt: Retransmission timeout reached on transmission a9s9gsc-qPxN00amC-J8tw.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response
[Apr 12 14:47:29] WARNING[28896]: chan_sip.c:4083 retrans_pkt: Hanging up call a9s9gsc-qPxN00amC-J8tw.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Apr 12 14:47:29]     -- Channel SIP/1_1_1_1-00000000 left 'simple_bridge' basic-bridge <39afb92d-e7da-411b-8251-eea995876826>
[Apr 12 14:47:29]   == Spawn extension (context_tok, 05052424142, 2) exited non-zero on 'SIP/1_1_1_1-00000000'
[Apr 12 14:47:29] Scheduling destruction of SIP dialog 'a9s9gsc-qPxN00amC-J8tw..' in 6400 ms (Method: INVITE)
[Apr 12 14:47:29] Reliably Transmitting (NAT) to 111.108.30.208:61654:
[Apr 12 14:47:29] BYE sip:1_1_1_1@111.108.30.208:60463;transport=UDP SIP/2.0
[Apr 12 14:47:29] Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK3a46278e;rport
[Apr 12 14:47:29] Max-Forwards: 70
[Apr 12 14:47:29] From: <sip:05052424142@52.194.253.25:15060;transport=UDP>;tag=as514844ef
[Apr 12 14:47:29] To: <sip:1_1_1_1@52.194.253.25:15060;transport=UDP>;tag=a93b3860
[Apr 12 14:47:29] Call-ID: a9s9gsc-qPxN00amC-J8tw..
[Apr 12 14:47:29] CSeq: 102 BYE
[Apr 12 14:47:29] User-Agent: Asterisk PBX 13.11.0-rc1
[Apr 12 14:47:29] Proxy-Authorization: Digest username="1_1_1_1", realm="asterisk", algorithm=MD5, uri="sip:52.194.253.25", nonce="4b615513", response="f7da3428126edeb1570a8935002af463"
[Apr 12 14:47:29] X-Asterisk-HangupCause: No user responding
[Apr 12 14:47:29] X-Asterisk-HangupCauseCode: 18
[Apr 12 14:47:29] Content-Length: 0
[Apr 12 14:47:29] 
[

Thank you in advance.

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#2

Your log is incomplete, so it is not certain whether or not this is an incoming re-invite, but it is certainly an incoming, not an outgoing, transaction that is failing…

You need to provide all INVITEs and their responses, for the session. You don’t seem to have any relevant INVITEs.

However, i’d like to know why you are using a non-standard SIP port number of 15060.

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#3

I’m doing in same way…good practice to hide yourself if you going outsaide of LAN.

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#4

Very good method beside IPTAbLES and some other firewall rules to avoid hacking attempts, changing the signaling port wont cause any issue on the SIP session if traffic can flow properly between the peers involved on the calls. It is something I hav done many times

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#5

these are errors are related to transmission of packets . Packet is sent to the destination and there is no response from the other side either it could be because of the firewall or the host has some problem.

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#6

If you re on a natted network you should uncomment this line, also add the localnet parameter and the careinvite it is depracated in recent versions of Asterisk use directmedia=no

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#7

Also this call was hangup due to the mising ACK " And according to the RFC3261, any SIP device not receiving the ACK to its final 2xx reply has to disconnect the call by issuing a standard BYE request.

I shared a post about this issue in a previous

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#8

The nat= options you quote are intended for the case where the peer is inside NAT and Asterisk is on a public address.

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#9

I made reference to NAT setting because there is private IP on the IP address for RTP stream

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#10

That only needs comedia.

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#11

If need comedia is pretty much different to never

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#12

Actually, if Asterisk is sending it in the SDP, comedia won’t help; the equivalent of comedia needs to be set on the peer, not Asteirsk.

In this case, comedia does seem to be needed, but that is not because Asterisk is natted, but because the peer is natted. In that case all bets are off on force_rport, but there is insufficient log to tell.

Peer - NAT - cloud - NAT - Asterisk

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#13

Well based on what you have define here seem now that is the best option try what I said before nat = force_rport,comedia

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#14

This wasn’t a simple NAT case, but my concern is there is a tendancy to assume that nat=yes should be used for all NAT cases, when it is actually intended for less common cases. Also, it seems to me that the reason for deprecating nat=yes was that people were using it without thinking, but it seems they are still doing that by using force_rport,comedia, because that is what they are told “yes” equates to.

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#15

Guys,

First of all, Thank you for the many responses.
It’s really higly appriciated and I love learning more about Asterisk.

It is such a great project.
I hope to help more people in the future also with the learning process right now.

Give me some time to review your answers, read more on the net and test some things.
I will get back to you.

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