Incoming Calls 404 Not found | Plzzz Help!

a2zit · July 10, 2016, 4:38pm

Recently configured PBX, outgoing calls works fine, but incoming calls is not working at all, i see a 404 not found SIP error in logs.

Logs:

<— SIP read from UDP:127.0.0.1:5060 —>
INVITE sip:+13214221387@192.168.1.123:5060 SIP/2.0
Record-Route: sip:127.0.0.1;r2=on;lr=on;ftag=sansay3756499272rdb6165;vsf=eBoZSnNfGX1WXl5xBwUfAHMAGh8GABgdADczLjMx;nat=yes
Record-Route: sip:192.168.1.123;r2=on;lr=on;ftag=sansay3756499272rdb6165;vsf=eBoZSnNfGX1WXl5xBwUfAHMAGh8GABgdADczLjMx;nat=yes
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0b3c.b39f6095aad896ef78838ef8f161ef55.0
Via: SIP/2.0/UDP 64.136.173.31:5060;rport=5060;branch=z9hG4bK1sansay3756499272rdb6165
Record-Route: sip:sansay3756499272rdb6165@64.136.173.31:5060;lr;transport=udp
To: sip:+13214221387@192.168.1.123:5060
From: sip:In-Main@64.136.173.31;tag=sansay3756499272rdb6165
Call-ID: 1195088237-0-1048481728@64.136.173.225
CSeq: 1 INVITE
Contact: sip:14077011111@64.136.173.31:5060
Supported: timer
Session-Expires: 1800;refresher=uac
Min-SE: 90
P-Asserted-Identity: sip:+14077011111@4.55.9.227:5060
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 319

v=0
o=Sansay-VSXi 188 1 IN IP4 192.168.1.123
s=Session Controller
c=IN IP4 192.168.1.123
t=0 0
m=audio 10944 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
a=nortpproxy:yes
<------------->
— (18 headers 15 lines) —
Sending to 127.0.0.1:5060 (NAT)
Sending to 127.0.0.1:5060 (NAT)
Using INVITE request as basis request - 1195088237-0-1048481728@64.136.173.225
Found peer ‘In-Main’ for ‘In-Main’ from 127.0.0.1:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.123:10944
Looking for +13214221387 in from-trunk (domain 192.168.1.123)

<— Reliably Transmitting (NAT) to 127.0.0.1:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0b3c.b39f6095aad896ef78838ef8f161ef55.0;received=127.0.0.1;rport=5060
Via: SIP/2.0/UDP 64.136.173.31:5060;rport=5060;branch=z9hG4bK1sansay3756499272rdb6165
From: sip:In-Main@64.136.173.31;tag=sansay3756499272rdb6165
To: sip:+13214221387@192.168.1.123:5060;tag=as6dfcd817
Call-ID: 1195088237-0-1048481728@64.136.173.225
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1195088237-0-1048481728@64.136.173.225’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:127.0.0.1:5060 —>
ACK sip:+13214221387@192.168.1.123:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0b3c.b39f6095aad896ef78838ef8f161ef55.0
To: sip:+13214221387@192.168.1.123:5060;tag=as6dfcd817
From: sip:In-Main@64.136.173.31;tag=sansay3756499272rdb6165
Call-ID: 1195088237-0-1048481728@64.136.173.225
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

Rmcgrath · July 10, 2016, 5:41pm

Check your settings, on trace it seems send package to “Sending to 127.0.0.1:5060 (NAT)”

david551 · July 10, 2016, 5:59pm

Please provide your logs as unformatted text, so that content in angle brackets is not lost.

Please provide all of your dialplan for the from-trunk context. (If you didn’t write it, please redirect this to whoever did.)

a2zit · July 10, 2016, 9:05pm

I am not sure where is the NAT setting to be checked !

here is the log:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [+13214221387@from-pstn:1] Set(“SIP/In-Main-00000041”, “__FROM_DID=+13214221387”) in new stack
– Executing [+13214221387@from-pstn:2] NoOp(“SIP/In-Main-00000041”, “Received an unknown call with DID set to +13214221387”) in new stack
– Executing [+13214221387@from-pstn:3] Goto(“SIP/In-Main-00000041”, “s,a2”) in new stack
– Goto (from-pstn,s,2)
– Executing [s@from-pstn:2] Answer(“SIP/In-Main-00000041”, “”) in new stack
> 0x9c07d18 – Probation passed - setting RTP source address to 127.0.0.1:11186
– Executing [s@from-pstn:3] Wait(“SIP/In-Main-00000041”, “2”) in new stack
> 0x9c07d18 – Probation passed - setting RTP source address to 127.0.0.1:11186
– Executing [s@from-pstn:4] Playback(“SIP/In-Main-00000041”, “ss-noservice”) in new stack
– <SIP/In-Main-00000041> Playing ‘ss-noservice.gsm’ (language ‘en’)
== Spawn extension (from-pstn, s, 4) exited non-zero on ‘SIP/In-Main-00000041’
– Executing [h@from-pstn:1] Hangup(“SIP/In-Main-00000041”, “”) in new stack
== Spawn extension (from-pstn, h, 1) exited non-zero on 'SIP/In-Main-00000041’
localhost*CLI> sip set debug on
SIP Debugging enabled

<— SIP read from UDP:127.0.0.1:5060 —>
INVITE sip:+13214221387@192.168.1.123:5060 SIP/2.0
Record-Route: sip:127.0.0.1;r2=on;lr=on;ftag=sansay3757395905rdb22368;vsf=eBoZSnNfGX1WXl5xBwUfAHMAGh8GABgdADczLjMx;nat=yes
Record-Route: sip:192.168.1.123;r2=on;lr=on;ftag=sansay3757395905rdb22368;vsf=eBoZSnNfGX1WXl5xBwUfAHMAGh8GABgdADczLjMx;nat=yes
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK049.52f2d25c535a3676f9a18054f6ffbdba.0
Via: SIP/2.0/UDP 64.136.173.31:5060;rport=5060;branch=z9hG4bK1sansay3757395905rdb22368
Record-Route: sip:sansay3757395905rdb22368@64.136.173.31:5060;lr;transport=udp
To: sip:+13214221387@192.168.1.123:5060
From: sip:In-Main@64.136.173.31;tag=sansay3757395905rdb22368
Call-ID: 1195378179-0-1064498958@64.136.173.225
CSeq: 1 INVITE
Contact: sip:14077011111@64.136.173.31:5060
Supported: timer
Session-Expires: 1800;refresher=uac
Min-SE: 90
P-Asserted-Identity: sip:+14077011111@4.55.9.227:5060
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 319

v=0
o=Sansay-VSXi 188 1 IN IP4 192.168.1.123
s=Session Controller
c=IN IP4 192.168.1.123
t=0 0
m=audio 14180 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
a=nortpproxy:yes
<------------->
— (18 headers 15 lines) —
Sending to 127.0.0.1:5060 (NAT)
Sending to 127.0.0.1:5060 (NAT)
Using INVITE request as basis request - 1195378179-0-1064498958@64.136.173.225
Found peer ‘In-Main’ for ‘In-Main’ from 127.0.0.1:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.123:14180
Looking for +13214221387 in from-pstn (domain 192.168.1.123)
list_route: hop: sip:127.0.0.1;r2=on;lr=on;ftag=sansay3757395905rdb22368;vsf=eBoZSnNfGX1WXl5xBwUfAHMAGh8GABgdADczLjMx;nat=yes
list_route: hop: sip:192.168.1.123;r2=on;lr=on;ftag=sansay3757395905rdb22368;vsf=eBoZSnNfGX1WXl5xBwUfAHMAGh8GABgdADczLjMx;nat=yes
list_route: hop: sip:sansay3757395905rdb22368@64.136.173.31:5060;lr;transport=udp

<— Transmitting (NAT) to 127.0.0.1:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK049.52f2d25c535a3676f9a18054f6ffbdba.0;received=127.0.0.1;rport=5060
Via: SIP/2.0/UDP 64.136.173.31:5060;rport=5060;branch=z9hG4bK1sansay3757395905rdb22368
Record-Route: sip:127.0.0.1;r2=on;lr=on;ftag=sansay3757395905rdb22368;vsf=eBoZSnNfGX1WXl5xBwUfAHMAGh8GABgdADczLjMx;nat=yes
Record-Route: sip:192.168.1.123;r2=on;lr=on;ftag=sansay3757395905rdb22368;vsf=eBoZSnNfGX1WXl5xBwUfAHMAGh8GABgdADczLjMx;nat=yes
Record-Route: sip:sansay3757395905rdb22368@64.136.173.31:5060;lr;transport=udp
From: sip:In-Main@64.136.173.31;tag=sansay3757395905rdb22368
To: sip:+13214221387@192.168.1.123:5060
Call-ID: 1195378179-0-1064498958@64.136.173.225
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: sip:+13214221387@127.0.0.1:5080
Content-Length: 0

<------------>
– Executing [+13214221387@from-pstn:1] Set(“SIP/In-Main-00000042”, “__FROM_DID=+13214221387”) in new stack
– Executing [+13214221387@from-pstn:2] NoOp(“SIP/In-Main-00000042”, “Received an unknown call with DID set to +13214221387”) in new stack
– Executing [+13214221387@from-pstn:3] Goto(“SIP/In-Main-00000042”, “s,a2”) in new stack
– Goto (from-pstn,s,2)
– Executing [s@from-pstn:2] Answer(“SIP/In-Main-00000042”, “”) in new stack
Audio is at 13024
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 127.0.0.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK049.52f2d25c535a3676f9a18054f6ffbdba.0;received=127.0.0.1;rport=5060
Via: SIP/2.0/UDP 64.136.173.31:5060;rport=5060;branch=z9hG4bK1sansay3757395905rdb22368
Record-Route: sip:127.0.0.1;r2=on;lr=on;ftag=sansay3757395905rdb22368;vsf=eBoZSnNfGX1WXl5xBwUfAHMAGh8GABgdADczLjMx;nat=yes
Record-Route: sip:192.168.1.123;r2=on;lr=on;ftag=sansay3757395905rdb22368;vsf=eBoZSnNfGX1WXl5xBwUfAHMAGh8GABgdADczLjMx;nat=yes
Record-Route: sip:sansay3757395905rdb22368@64.136.173.31:5060;lr;transport=udp
From: sip:In-Main@64.136.173.31;tag=sansay3757395905rdb22368
To: sip:+13214221387@192.168.1.123:5060;tag=as70bba852
Call-ID: 1195378179-0-1064498958@64.136.173.225
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: sip:+13214221387@127.0.0.1:5080
Content-Type: application/sdp
Require: timer
Content-Length: 299