Calls dropping after approx 6-8 seconds

Hi All,

I’m having an issue with a setup where inbound SIP calls from another PBX are dropping after approx 6-8 seconds. Calls from the asterisk box to the NEC PBX work fine.

I can see from the trace on the Asterisk server that once the call is established there is a continuous exchange between the asterisk server & the PBX in which asterisk sends a 200 OK message and the PBX replies with an ACK, but then the asterisk server re-transmits the OK ( See debug in next post)

For simplicity & security I have replaced the real IP & hostname of the A2billing server in the logs to: 9.9.9.9 (my.asterisk.server) and the NEC PBX IP to 5.5.5.5

Note that both servers pass through firewalls and are NATd. While i appreciate that NAT can be a problem with SIP I cannot remove NAT from the setup unfortunately. So if NAT is part of the problem here i’d need to look at what i can do to resolve the issue while keeping NAT in the picture.

I know this is a problem, but cannot see what i can do to resolve. Any assistance would be greatly appreciated.

E2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
INVITE sip:[email protected];transport=tcp SIP/2.0
From: sip:[email protected];tag=A2443246313536410009F6A0
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060;transport=tcp
Content-Type: application/sdp
Remote-Party-ID: sip:[email protected];party=calling;screen=no;privacy=full
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: [email protected]
CSeq: 1 INVITE
Route: sip:my.asterisk.server;lr
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK5F8006C6C7104D15
Content-Length: 304

v=0
o=- 0 0 IN IP4 5.5.5.5
s=T035
c=IN IP4 5.5.5.5
t=0 0
m=audio 10020 RTP/AVP 8 2 18 9 110
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:9 G722/8000
a=ptime:30
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
<------------->

e[KE2-SIP-01*CLI>
e[0K— (16 headers 16 lines) —
Sending to 5.5.5.5:5060 (NAT)
Using INVITE request as basis request - [email protected]

e[KE2-SIP-01*CLI>
e[0KFound peer ‘0744461921’ for ‘anonymous’ from 5.5.5.5:5060

<— Reliably Transmitting (NAT) to 5.5.5.5:5060 —>
SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK5F8006C6C7104D15;received=5.5.5.5;rport=5060From: sip:[email protected];tag=A2443246313536410009F6A0To: sip:[email protected]:5060;tag=as4bf6427aCall-ID: [email protected]: 1 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerWWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4928fcee"Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 8576 ms (Method: INVITE)

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
ACK sip:[email protected];transport=tcp SIP/2.0
Call-ID: [email protected]
CSeq: 1 ACK
From: sip:[email protected];tag=A2443246313536410009F6A0
To: sip:[email protected]:5060;tag=as4bf6427a
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK5F8006C6C7104D15
Max-Forwards: 70
Route: sip:my.asterisk.server;lr
User-Agent: NEC-i SV8100-GE 08.00
Content-Length: 0

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

e[KE2-SIP-01*CLI>
e[0K – SIP/SOUL_OUTBOUND-0000dbdb is ringing

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
INVITE sip:[email protected];transport=tcp SIP/2.0
From: sip:[email protected];tag=A2653246313536410009F6A3
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060;transport=tcp
Content-Type: application/sdp
Remote-Party-ID: sip:[email protected];party=calling;screen=no;privacy=full
CSeq: 2 INVITE
Authorization: Digest username=“0744461921”,realm=“asterisk”,algorithm=MD5,nonce=“4928fcee”,opaque="",uri=“sip:[email protected]”,response="0027de822c0886593adc7b6415bbb60a"
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: [email protected]
Route: sip:my.asterisk.server;lr
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818
Content-Length: 304

v=0
o=- 0 0 IN IP4 5.5.5.5
s=T035
c=IN IP4 5.5.5.5
t=0 0
m=audio 10020 RTP/AVP 8 2 18 9 110
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:9 G722/8000
a=ptime:30
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
<------------->

e[KE2-SIP-01*CLI>
e[0K— (17 headers 16 lines) —
Sending to 5.5.5.5:5060 (NAT)
Using INVITE request as basis request - [email protected]

e[KE2-SIP-01*CLI>
e[0KFound peer ‘0744461921’ for ‘anonymous’ from 5.5.5.5:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 9

e[KE2-SIP-01*CLI>
e[0KFound RTP audio format 110
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 110
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x1908 (alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x108 (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

e[KE2-SIP-01*CLI>
e[0KPeer audio RTP is at port 5.5.5.5:10020
Looking for 0386969318 in empower-private (domain my.asterisk.server)
list_route: hop: sip:[email protected]:5060;transport=tcp

e[KE2-SIP-01*CLI>
e[0K
<— Transmitting (NAT) to 5.5.5.5:5060 —>
SIP/2.0 100 TryingVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: sip:[email protected];tag=A2653246313536410009F6A3To: sip:[email protected]:5060Call-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: sip:[email protected]:5060Content-Length: 0
> Limit Data for this call:

e[KE2-SIP-01*CLI>
e[0K > timelimit = 7200000 ms (7200.000 s)
> play_warning = 61000 ms (61.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000 ms (30.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =

e[KE2-SIP-01*CLI>
e[0K == Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

e[KE2-SIP-01*CLI>
e[0K – Called SIP/SOUL_OUTBOUND/0386969318

<— Transmitting (NAT) to 5.5.5.5:5060 —>
SIP/2.0 180 RingingVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: sip:[email protected];tag=A2653246313536410009F6A3To: sip:[email protected]:5060;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: sip:[email protected]:5060Content-Length: 0
<------------>

e[KE2-SIP-01*CLI>
e[0K – SIP/SOUL_OUTBOUND-0000dbdb is ringing

e[KE2-SIP-01*CLI>
e[0K – SIP/SOUL_OUTBOUND-0000dbdb answered SIP/0884936015-0000dbda

e[KE2-SIP-01*CLI>
e[0K – SIP/SOUL_OUTBOUND-0000dbdd is ringing

<— Transmitting (NAT) to 5.5.5.5:5060 —>
SIP/2.0 180 RingingVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: sip:[email protected];tag=A2653246313536410009F6A3To: sip:[email protected]:5060;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: sip:[email protected]:5060Content-Length: 0
<------------>

e[KE2-SIP-01*CLI>
e[0K – SIP/SOUL_OUTBOUND-0000dbdd is making progress passing it to SIP/0744461921-0000dbdc

e[KE2-SIP-01*CLI>
e[0K – SIP/SOUL_OUTBOUND-0000dbdd answered SIP/0744461921-0000dbdc

e[KE2-SIP-01*CLI>
e[0KAudio is at 19900
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 5.5.5.5:5060 —>
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: sip:[email protected];tag=A2653246313536410009F6A3To: sip:[email protected]:5060;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: sip:[email protected]:5060Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110 telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv
<------------>

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: sip:[email protected]:5060;tag=as6df6ebba
From: sip:[email protected];tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

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

e[KE2-SIP-01*CLI>
e[0KRetransmitting #1 (NAT) to 5.5.5.5:5060:
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: sip:[email protected];tag=A2653246313536410009F6A3To: sip:[email protected]:5060;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: sip:[email protected]:5060Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110 telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: sip:[email protected]:5060;tag=as6df6ebba
From: sip:[email protected];tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

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

e[KE2-SIP-01*CLI>
e[0KRetransmitting #2 (NAT) to 5.5.5.5:5060:
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: sip:[email protected];tag=A2653246313536410009F6A3To: sip:[email protected]:5060;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: sip:[email protected]:5060Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110 telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: sip:[email protected]:5060;tag=as6df6ebba
From: sip:[email protected];tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

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

e[KE2-SIP-01*CLI>
e[0KRetransmitting #3 (NAT) to 5.5.5.5:5060:
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: sip:[email protected];tag=A2653246313536410009F6A3To: sip:[email protected]:5060;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: sip:03869693[email protected]:5060Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110 telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: sip:[email protected]:5060;tag=as6df6ebba
From: sip:[email protected];tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

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

e[KE2-SIP-01*CLI>
e[0KRetransmitting #4 (NAT) to 5.5.5.5:5060:
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: sip:[email protected];tag=A2653246313536410009F6A3To: sip:[email protected]:5060;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: sip:[email protected]:5060Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110 telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: sip:[email protected]:5060;tag=as6df6ebba
From: sip:[email protected];tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

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

e[KE2-SIP-01*CLI>
e[0K == Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> Limit Data for this call:
> timelimit = 7200000 ms (7200.000 s)
> play_warning = 61000 ms (61.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000 ms (30.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =

e[KE2-SIP-01*CLI>
e[0K == Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

e[KE2-SIP-01*CLI>
e[0K – Called SIP/SOUL_OUTBOUND/0396207531

e[KE2-SIP-01*CLI>
e[0K – SIP/SOUL_OUTBOUND-0000dbdf is ringing

e[KE2-SIP-01*CLI>
e[0K – SIP/SOUL_OUTBOUND-0000dbdf is making progress passing it to SIP/2657549926-0000dbde

e[KE2-SIP-01*CLI>
e[0KRetransmitting #5 (NAT) to 5.5.5.5:5060:
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: sip:[email protected];tag=A2653246313536410009F6A3To: sip:[email protected]:5060;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: sip:[email protected]:5060Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110 telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: sip:[email protected]:5060;tag=as6df6ebba
From: sip:[email protected];tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

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

e[KE2-SIP-01*CLI>
e[0KRetransmitting #6 (NAT) to 5.5.5.5:5060:
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: sip:[email protected];tag=A2653246313536410009F6A3To: sip:[email protected]:5060;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: sip:[email protected]:5060Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110 telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: sip:[email protected]:5060;tag=as6df6ebba
From: sip:[email protected];tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

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

e[KE2-SIP-01*CLI>
e[0KReliably Transmitting (NAT) to 5.5.5.5:5060:
BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rportMax-Forwards: 70From: sip:[email protected]:5060;tag=as6df6ebbaTo: sip:[email protected];tag=A2653246313536410009F6A3Call-ID: [email protected]: 102 BYEUser-Agent: FPBX-2.8.1 (1.8.15.0)Proxy-Authorization: Digest username=“0744461921”, realm=“asterisk”, algorithm=MD5, uri=“sip:my.asterisk.server”, nonce="", response="963caa5a84822bd1da888a3022080ae3"X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
SIP/2.0 200 OK
From: sip:[email protected]:5060;tag=as6df6ebba
To: sip:[email protected];tag=A2653246313536410009F6A3
Call-ID: [email protected]
CSeq: 102 BYE
Server: NEC-i SV8100-GE 08.00/2.1
Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rport
Content-Length: 0

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

e[KE2-SIP-01*CLI>
e[0KRetransmitting #1 (NAT) to 5.5.5.5:5060:
BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rportMax-Forwards: 70From: sip:[email protected]:5060;tag=as6df6ebbaTo: sip:[email protected];tag=A2653246313536410009F6A3Call-ID: [email protected]: 102 BYEUser-Agent: FPBX-2.8.1 (1.8.15.0)Proxy-Authorization: Digest username=“0744461921”, realm=“asterisk”, algorithm=MD5, uri=“sip:my.asterisk.server”, nonce="", response="963caa5a84822bd1da888a3022080ae3"X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
SIP/2.0 200 OK
From: sip:[email protected]:5060;tag=as6df6ebba
To: sip:[email protected];tag=A2653246313536410009F6A3
Call-ID: [email protected]
CSeq: 102 BYE
Server: NEC-i SV8100-GE 08.00/2.1
Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rport
Content-Length: 0

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

e[KE2-SIP-01*CLI>
e[0KRetransmitting #2 (NAT) to 5.5.5.5:5060:
BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rportMax-Forwards: 70From: sip:[email protected]:5060;tag=as6df6ebbaTo: sip:[email protected];tag=A2653246313536410009F6A3Call-ID: [email protected]: 102 BYEUser-Agent: FPBX-2.8.1 (1.8.15.0)Proxy-Authorization: Digest username=“0744461921”, realm=“asterisk”, algorithm=MD5, uri=“sip:my.asterisk.server”, nonce="", response="963caa5a84822bd1da888a3022080ae3"X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
SIP/2.0 200 OK
From: sip:[email protected]:5060;tag=as6df6ebba
To: sip:[email protected];tag=A2653246313536410009F6A3
Call-ID: [email protected]
CSeq: 102 BYE
Server: NEC-i SV8100-GE 08.00/2.1
Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rport
Content-Length: 0

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

e[KE2-SIP-01*CLI>
e[0KRetransmitting #3 (NAT) to 5.5.5.5:5060:
BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rportMax-Forwards: 70From: sip:[email protected]:5060;tag=as6df6ebbaTo: sip:[email protected];tag=A2653246313536410009F6A3Call-ID: [email protected]: 102 BYEUser-Agent: FPBX-2.8.1 (1.8.15.0)Proxy-Authorization: Digest username=“0744461921”, realm=“asterisk”, algorithm=MD5, uri=“sip:my.asterisk.server”, nonce="", response="963caa5a84822bd1da888a3022080ae3"X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
SIP/2.0 200 OK
From: sip:[email protected]:5060;tag=as6df6ebba
To: sip:[email protected];tag=A2653246313536410009F6A3
Call-ID: [email protected]
CSeq: 102 BYE
Server: NEC-i SV8100-GE 08.00/2.1
Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rport
Content-Length: 0

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

e[KE2-SIP-01*CLI>
e[0KRetransmitting #4 (NAT) to 5.5.5.5:5060:
BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rportMax-Forwards: 70From: sip:[email protected]:5060;tag=as6df6ebbaTo: sip:[email protected];tag=A2653246313536410009F6A3Call-ID: [email protected]: 102 BYEUser-Agent: FPBX-2.8.1 (1.8.15.0)Proxy-Authorization: Digest username=“0744461921”, realm=“asterisk”, algorithm=MD5, uri=“sip:my.asterisk.server”, nonce="", response="963caa5a84822bd1da888a3022080ae3"X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
SIP/2.0 200 OK
From: sip:[email protected]:5060;tag=as6df6ebba
To: sip:[email protected];tag=A2653246313536410009F6A3
Call-ID: [email protected]
CSeq: 102 BYE
Server: NEC-i SV8100-GE 08.00/2.1
Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rport
Content-Length: 0

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

e[KE2-SIP-01*CLI>
e[0KRetransmitting #5 (NAT) to 5.5.5.5:5060:
BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rportMax-Forwards: 70From: sip:[email protected]:5060;tag=as6df6ebbaTo: sip:[email protected];tag=A2653246313536410009F6A3Call-ID: [email protected]: 102 BYEUser-Agent: FPBX-2.8.1 (1.8.15.0)Proxy-Authorization: Digest username=“0744461921”, realm=“asterisk”, algorithm=MD5, uri=“sip:my.asterisk.server”, nonce="", response="963caa5a84822bd1da888a3022080ae3"X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0

e[KE2-SIP-01*CLI>
e[0K
<— SIP read from UDP:5.5.5.5:5060 —>
SIP/2.0 200 OK
From: sip:[email protected]:5060;tag=as6df6ebba
To: sip:[email protected];tag=A2653246313536410009F6A3
Call-ID: [email protected]
CSeq: 102 BYE
Server: NEC-i SV8100-GE 08.00/2.1
Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rport
Content-Length: 0

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

e[KE2-SIP-01*CLI> quite[0me]0;[email protected]: [email protected]:~$

Do you have 5.5.5.5 in your SIP localnet settings so the NEC is excluded from NAT processing?

Hi Skyking,

No i don’t, the 2 systems are remote from one another and are NAT’d through their respective firewalls. (See diagram below)

ASTERISK << >> FIREWALL << INTERNET >> FIREWALL << >> NEC PBX

The 5.5.5.5 & 9.9.9.9 addresses in the debugs are the public NAT’d addresses.

Regards

Something doesn’t support NAT. Asterisk can’t establish media continuity so it is clearing the call. I would use a VPN.

HI,

I wish that was an option, but it is not. The Asterisk server is our ITSP SIP server, and therefore connects to numerous customers many of whom have clashing subnets. So i cannot utilise VPN’s or private connections.

I’m more looking to see what in particular aspect of the process is failing and why, and then seeing if i can address that.

I’ve got numerous IP PBX systems working with NAT in the middle, just never a NEC PBX; so if i can identify the root cause i might be able to modify the NEC’s behavior to work around this.

I’ve been working with NEC support and they have provided guides for ITSP providers ( all of which involve NAT) so i know that the NEC can work in this scenario too.

Regards

Also please note that i have 2 way audio; so the RTP stream is OK.

Please elucidate on your two “FIREWALL(s)” a common mistake is to rely on SIP “helpers” that don’t, the SIP session is best left “unhelped” especially in a double natted environment, just forward the traffic.

Hi Dicko,

If i’m correct in thinking that you are referring to SIP inspection/ALG, then this has not been enabled on either firewall.

As you point out it can cause problems. Both of the SIP systems are already setting the appropriate IPs or hostnames in the SIP messages, so the firewalls have been setup to leave the SIP messages alone.

Regards

Yet something is ignoring or overriding the timers, or the SIP transactions are being ignored, no?

set the sip debug “on” to your questioned IP on both endpoints to debug maybe. . .

This is my assumption, but i’m not sure where exactly the issues lies ( Asterisk, the firewalls, NEC PBX)

The debug from the asterisk server is at the start of my post. I thought i’d start the ball rolling there.

I’m arranging to capture the debug from the NEC, i need to setup port mirroring and wireshark for the NEC so it’s a bit more involved.

It’s a common mistake, thinking that because you have 2 way audio all is OK. Asterisk is very picky. I am sure you will see in the log that Asterisk can’t see the RTP session associated with the SIP channel. Perhaps your phone is inviting the audio directly, bypassing Asterisk. Asterisk RTP debug would answer this.

Also some systems are very confused by the connected line/RPID updates. This can also cause the media to try and anchor off the call. The RPID comes about 6 seconds after establishment. I would watch both the SIP and RTP debug (not both at once, too confusing, and turn verbosity to 0) and the culprit should be readily apparent.

Thanks for the idea. I’ve debugged RTP and can see RTP traffic in both directions. Based on the SIP debugs i believe the issue is related to re-invites. I have tried setting my canreinvite value to YES, NO, UPDATE. I’m going to be getting atrace from the NEC tomorrow so will be able to look at that. I’ll ask check with NEC Support on their reinvite support/options.