404 not found

system · October 10, 2014, 2:03am

Hi.

This is my first post in this community so forgive me if this isn’t the right topic for my problem.

When someone is calling a none existing phone number I receive a 404 from my provider and I think it’s the right code.

The user experience is a bit odd because when dialing you just get a silence for about 5 seconds and then the tone matching a busy signal. In Denmark we would like to have a signal matching the tone on this site: http://en.wikipedia.org/wiki/Special_information_tones - play tone Intercept.

Does anyone have an idea how to send an intercept tone?

You find the SIP-debug dump here. My provider is x.x.x.x and my Asterisk at the edge is y.y.y.y and z.z.z.z is the Asterisk where clients are connected.

Reliably Transmitting (no NAT) to x.x.x.x:5060:
OPTIONS sip:x.x.x.x SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK44338242
Max-Forwards: 70
From: "asterisk" <sip:asterisk@y.y.y.y>;tag=as6bf9aa6b
To: <sip:x.x.x.x>
Contact: <sip:asterisk@y.y.y.y:5060>
Call-ID: 2a582a2f75861b597ecce63800d2be13@y.y.y.y:5060
CSeq: 102 OPTIONS
User-Agent: MY-SERVER AST-TC-01
Date: Fri, 10 Oct 2014 00:53:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:x.x.x.x:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK44338242;received=y.y.y.y
From: "asterisk" <sip:asterisk@y.y.y.y>;tag=as6bf9aa6b
To: <sip:x.x.x.x>;tag=as3e2595f1
Call-ID: 2a582a2f75861b597ecce63800d2be13@y.y.y.y:5060
CSeq: 102 OPTIONS
Server: Provider Wholesale
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:x.x.x.x:5060>
Accept: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '2a582a2f75861b597ecce63800d2be13@y.y.y.y:5060' Method: OPTIONS

<--- SIP read from UDP:z.z.z.z:5060 --->
INVITE sip:41420502@y.y.y.y:5060 SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK708c6094
Max-Forwards: 70
To: <sip:41420502@y.y.y.y:5060>
Contact: <sip:88336633@z.z.z.z:5060>
Call-ID: 56e8527d3acaecd23f2ecbf52efab05c@z.z.z.z:5060
CSeq: 102 INVITE
User-Agent: MY-SERVER
Date: Fri, 10 Oct 2014 00:53:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Privacy: off
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1584925282 1584925282 IN IP4 z.z.z.z
s=Asterisk PBX 11.4.0
c=IN IP4 z.z.z.z
t=0 0
m=audio 17374 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to z.z.z.z:5060 (no NAT)
Sending to z.z.z.z:5060 (no NAT)
Using INVITE request as basis request - 56e8527d3acaecd23f2ecbf52efab05c@z.z.z.z:5060
Found peer 'MC01' for '88336633' from z.z.z.z:5060

<--- Reliably Transmitting (no NAT) to z.z.z.z:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK708c6094;received=z.z.z.z
From: "MY-SERVER" <sip:88336633@z.z.z.z>;tag=as00bd5e60
To: <sip:41420502@y.y.y.y:5060>;tag=as5a4fc946
Call-ID: 56e8527d3acaecd23f2ecbf52efab05c@z.z.z.z:5060
CSeq: 102 INVITE
Server: MY-SERVER AST-TC-01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="134b1268"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '56e8527d3acaecd23f2ecbf52efab05c@z.z.z.z:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:z.z.z.z:5060 --->
ACK sip:41420502@y.y.y.y:5060 SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK708c6094
Max-Forwards: 70
From: "MY-SERVER" <sip:88336633@z.z.z.z>;tag=as00bd5e60
To: <sip:41420502@y.y.y.y:5060>;tag=as5a4fc946
Contact: <sip:88336633@z.z.z.z:5060>
Call-ID: 56e8527d3acaecd23f2ecbf52efab05c@z.z.z.z:5060
CSeq: 102 ACK
User-Agent: MY-SERVER
Content-Length: 0

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

<--- SIP read from UDP:z.z.z.z:5060 --->
INVITE sip:41420502@y.y.y.y:5060 SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK33f4ee7e
Max-Forwards: 70
From: "MY-SERVER" <sip:88336633@z.z.z.z>;tag=as00bd5e60
To: <sip:41420502@y.y.y.y:5060>
Contact: <sip:88336633@z.z.z.z:5060>
Call-ID: 56e8527d3acaecd23f2ecbf52efab05c@z.z.z.z:5060
CSeq: 103 INVITE
User-Agent: MY-SERVER
Authorization: Digest username="MC01", realm="asterisk", algorithm=MD5, uri="sip:41420502@y.y.y.y:5060", nonce="134b1268", response="f4bd374afcd0d9a88a25cb8ae30779ac"
Date: Fri, 10 Oct 2014 00:53:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Privacy: off
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1584925282 1584925283 IN IP4 z.z.z.z
s=Asterisk PBX 11.4.0
c=IN IP4 z.z.z.z
t=0 0
m=audio 17374 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (16 headers 12 lines) ---
Sending to z.z.z.z:5060 (no NAT)
Using INVITE request as basis request - 56e8527d3acaecd23f2ecbf52efab05c@z.z.z.z:5060
Found peer 'MC01' for '88336633' from z.z.z.z:5060
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port z.z.z.z:17374
Looking for 41420502 in Extern_Dialplan (domain y.y.y.y)
list_route: hop: <sip:88336633@z.z.z.z:5060>

<--- Transmitting (no NAT) to z.z.z.z:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK33f4ee7e;received=z.z.z.z
From: "MY-SERVER" <sip:88336633@z.z.z.z>;tag=as00bd5e60
To: <sip:41420502@y.y.y.y:5060>
Call-ID: 56e8527d3acaecd23f2ecbf52efab05c@z.z.z.z:5060
CSeq: 103 INVITE
Server: MY-SERVER AST-TC-01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:41420502@y.y.y.y:5060>
Content-Length: 0


<------------>
Audio is at 11138
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to x.x.x.x:5060:
INVITE sip:41420502@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK3aea3073
Max-Forwards: 70
From: "MY-SERVER" <sip:88336633@y.y.y.y>;tag=as222e3a22
To: <sip:41420502@x.x.x.x:5060>
Contact: <sip:88336633@y.y.y.y:5060>
Call-ID: 44f0a98b37aafa013b53f42a2464552b@y.y.y.y:5060
CSeq: 102 INVITE
User-Agent: MY-SERVER AST-TC-01
Date: Fri, 10 Oct 2014 00:53:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Privacy: off
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 1917735604 1917735604 IN IP4 y.y.y.y
s=Asterisk PBX 11.12.1
c=IN IP4 y.y.y.y
t=0 0
m=audio 11138 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:x.x.x.x:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK3aea3073;received=y.y.y.y;rport=5060
From: "MY-SERVER" <sip:88336633@y.y.y.y>;tag=as222e3a22
To: <sip:41420502@x.x.x.x:5060>
Call-ID: 44f0a98b37aafa013b53f42a2464552b@y.y.y.y:5060
CSeq: 102 INVITE
Server: Provider Wholesale
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:41420502@x.x.x.x:5060>
Content-Length: 0

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

<--- SIP read from UDP:z.z.z.z:5060 --->
OPTIONS sip:y.y.y.y SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK77af9a81
Max-Forwards: 70
From: "asterisk" <sip:asterisk@z.z.z.z>;tag=as31946fdd
To: <sip:y.y.y.y>
Contact: <sip:asterisk@z.z.z.z:5060>
Call-ID: 0a58954d2e5b7c2d3e8f9f217bf12be5@z.z.z.z:5060
CSeq: 102 OPTIONS
User-Agent: MY-SERVER
Date: Fri, 10 Oct 2014 00:53:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to z.z.z.z:5060 (no NAT)
Looking for s in default (domain y.y.y.y)

<--- Transmitting (no NAT) to z.z.z.z:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK77af9a81;received=z.z.z.z
From: "asterisk" <sip:asterisk@z.z.z.z>;tag=as31946fdd
To: <sip:y.y.y.y>;tag=as5dc2b000
Call-ID: 0a58954d2e5b7c2d3e8f9f217bf12be5@z.z.z.z:5060
CSeq: 102 OPTIONS
Server: MY-SERVER AST-TC-01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0a58954d2e5b7c2d3e8f9f217bf12be5@z.z.z.z:5060' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '77f8c4bb092a5d43583d2e1974673fb7@x.x.x.x:5060' Method: OPTIONS

<--- SIP read from UDP:x.x.x.x:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK3aea3073;received=y.y.y.y;rport=5060
From: "MY-SERVER" <sip:88336633@y.y.y.y>;tag=as222e3a22
To: <sip:41420502@x.x.x.x:5060>;tag=as1cbba1a6
Call-ID: 44f0a98b37aafa013b53f42a2464552b@y.y.y.y:5060
CSeq: 102 INVITE
Server: Provider Wholesale
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Reason: Q.850;cause=1
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to x.x.x.x:5060:
ACK sip:41420502@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK3aea3073
Max-Forwards: 70
From: "MY-SERVER" <sip:88336633@y.y.y.y>;tag=as222e3a22
To: <sip:41420502@x.x.x.x:5060>;tag=as1cbba1a6
Contact: <sip:88336633@y.y.y.y:5060>
Call-ID: 44f0a98b37aafa013b53f42a2464552b@y.y.y.y:5060
CSeq: 102 ACK
User-Agent: MY-SERVER AST-TC-01
Content-Length: 0


---
Scheduling destruction of SIP dialog '44f0a98b37aafa013b53f42a2464552b@y.y.y.y:5060' in 6400 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to z.z.z.z:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK33f4ee7e;received=z.z.z.z
From: "MY-SERVER" <sip:88336633@z.z.z.z>;tag=as00bd5e60
To: <sip:41420502@y.y.y.y:5060>;tag=as4a6ea76b
Call-ID: 56e8527d3acaecd23f2ecbf52efab05c@z.z.z.z:5060
CSeq: 103 INVITE
Server: MY-SERVER AST-TC-01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0


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

<--- SIP read from UDP:z.z.z.z:5060 --->
ACK sip:41420502@y.y.y.y:5060 SIP/2.0
Via: SIP/2.0/UDP z.z.z.z:5060;branch=z9hG4bK33f4ee7e
Max-Forwards: 70
From: "MY-SERVER" <sip:88336633@z.z.z.z>;tag=as00bd5e60
To: <sip:41420502@y.y.y.y:5060>;tag=as4a6ea76b
Contact: <sip:88336633@z.z.z.z:5060>
Call-ID: 56e8527d3acaecd23f2ecbf52efab05c@z.z.z.z:5060
CSeq: 103 ACK
User-Agent: MY-SERVER
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '56e8527d3acaecd23f2ecbf52efab05c@z.z.z.z:5060' Method: ACK
ast-tc-01*CLI> sip set debug off
SIP Debugging Disabled

Cheers

/Ron9

david55 · October 10, 2014, 9:21am

Six seconds suggests that the network is sending the SIT sequence as early media then sending the actual invalid number code. You need to call the Progress application and have a local device that supports early media, or you need to call Answer before you dial (not generally best practice, but maybe necessary here).

Normally, a switch that was going to send early media would and 183 Progress, with SDP, and I am not sure that Asterisk will accept early media without the far side SDP. You may have to wait for the 404, then based on${HANGUPCAUSE}, use Answer or Progress and then Playtones, after configuring the right SIT in indications.conf. The default SIT may well be right for NU, as it is intended as a telemarketer deterrent.

As a general note, especially when mentioning timing, it is best to take log extracts from the log file, not a screen scrape.