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Francesco
2009-Mar-20 08:52 UTC
[asterisk-users] Asterisk Realtime Configuration and 404 Extension not found

Hi to all the ML. I'm new here.
I start to use asterisk with realtime configuration, with pgsql
backend connected via odbc.
The connection between asterisk and pgsql works fine.
I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501.
Those are the records:

asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf;
 name |  host   |  type  | context | secret | defaultuser
------+---------+--------+---------+--------+-------------
 1401 | dynamic | friend | prova   | fra    | 1401
 1501 | dynamic | friend | prova   | 1501   | 1501

I create a table extensions_conf with this extensions:

id | context |   exten   | priority |  app   |    appdata
----+---------+-----------+----------+--------+----------------
 41 | prova   | _1[1-5]XX |        1 | Dial   | (SIP/${EXTEN})
 42 | prova   | _1[1-5]XX |        2 | Hangup |

the extconfig.conf is:

[settings]
extensions => odbc,dbasterisk,extensions_conf
sipusers => odbc,dbasterisk,sip_conf
sippeers => odbc,dbasterisk,sip_conf

and extensions.conf is:

[general]
static=yes
writeprotect=no
autofallthrough=yes

[prova]
switch => Realtime

I use x-lite for calling 1401 -> 1501 and vice-versa.
The result is "404 Not Found". The phones correctly register on
asterisk, and the table sip_conf register the userage, the ip ecc.

Setting "sip set debug on" on asterisk console, i obtain:

------------------8<---------------------------8<---------------------------8<---------------------------8<---------------------------8<---------
Via: SIP/2.0/UDP
10.44.3.153:5060;rport;branch=z9hG4bK5591DE8FE488A6D64C4AA159B97BBC36
From: 1501 <sip:1501 at 10.44.9.0>;tag=990832587
To: <sip:1401 at 10.44.9.0>
Contact: <sip:1501 at 10.44.3.153:5060>
Call-ID: 448AFE95-423A-A081-85F3-AD0D70D0FB4D at 10.44.3.153
CSeq: 60784 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 305

v=0
o=1501 503728492 503728510 IN IP4 10.44.3.153
s=X-Lite
c=IN IP4 10.44.3.153
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (11 headers 14 lines) ---
 == Using SIP RTP CoS mark 5
Sending to 10.44.3.153 : 5060 (NAT)
Using INVITE request as basis request -
448AFE95-423A-A081-85F3-AD0D70D0FB4D at 10.44.3.153
Found peer '1501' for '1501' from 10.44.3.153:5060

<--- Reliably Transmitting (no NAT) to 10.44.3.153:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.44.3.153:5060;branch=z9hG4bK5591DE8FE488A6D64C4AA159B97BBC36;received=10.44.3.153;rport=5060
From: 1501 <sip:1501 at 10.44.9.0>;tag=990832587
To: <sip:1401 at 10.44.9.0>;tag=as4b3729a3
Call-ID: 448AFE95-423A-A081-85F3-AD0D70D0FB4D at 10.44.3.153
CSeq: 60784 INVITE
Server: Asterisk PBX 1.6.1-rc1-issue14292
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="10.44.9.0",
nonce="4685095b"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog
'448AFE95-423A-A081-85F3-AD0D70D0FB4D at 10.44.3.153' in 32000 ms
(Method: INVITE)
itpbx01*CLI>
<--- SIP read from UDP:10.44.3.153:5060 --->
ACK sip:1401 at 10.44.9.0 SIP/2.0
Via: SIP/2.0/UDP
10.44.3.153:5060;rport;branch=z9hG4bK5591DE8FE488A6D64C4AA159B97BBC36
From: 1501 <sip:1501 at 10.44.9.0>;tag=990832587
To: <sip:1401 at 10.44.9.0>;tag=as4b3729a3
Contact: <sip:1501 at 10.44.3.153:5060>
Call-ID: 448AFE95-423A-A081-85F3-AD0D70D0FB4D at 10.44.3.153
CSeq: 60784 ACK
Max-Forwards: 70
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
itpbx01*CLI>
<--- SIP read from UDP:10.44.3.153:5060 --->
INVITE sip:1401 at 10.44.9.0 SIP/2.0
Via: SIP/2.0/UDP
10.44.3.153:5060;rport;branch=z9hG4bK26FAB8569057F005F91285F58781AD32
From: 1501 <sip:1501 at 10.44.9.0>;tag=990832587
To: <sip:1401 at 10.44.9.0>
Contact: <sip:1501 at 10.44.3.153:5060>
Call-ID: 448AFE95-423A-A081-85F3-AD0D70D0FB4D at 10.44.3.153
CSeq: 60785 INVITE
Authorization: Digest
username="1501",realm="10.44.9.0",nonce="4685095b",response="290748094c61098ad389aaaadafa825e",uri="sip:1401
at 10.44.9.0",algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 305

v=0
o=1501 503728492 503728510 IN IP4 10.44.3.153
s=X-Lite
c=IN IP4 10.44.3.153
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (12 headers 14 lines) ---
Sending to 10.44.3.153 : 5060 (NAT)
Using INVITE request as basis request -
448AFE95-423A-A081-85F3-AD0D70D0FB4D at 10.44.3.153
Found peer '1501' for '1501' from 10.44.3.153:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 10.44.3.153:8000
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 98
Found audio description format speex for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x60e
(gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing),
combined - 0x40e (gsm|ulaw|alaw|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.44.3.153:8000
Looking for 1401 in prova (domain 10.44.9.0)
itpbx01*CLI>
<--- Reliably Transmitting (no NAT) to 10.44.3.153:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
10.44.3.153:5060;branch=z9hG4bK26FAB8569057F005F91285F58781AD32;received=10.44.3.153;rport=5060
From: 1501 <sip:1501 at 10.44.9.0>;tag=990832587
To: <sip:1401 at 10.44.9.0>;tag=as4b3729a3
Call-ID: 448AFE95-423A-A081-85F3-AD0D70D0FB4D at 10.44.3.153
CSeq: 60785 INVITE
Server: Asterisk PBX 1.6.1-rc1-issue14292
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 19 10:19:52] NOTICE[31891]: chan_sip.c:19476
handle_request_invite: Call from '1501' to extension '1401'
rejected
because extension not found.
Scheduling destruction of SIP dialog
'448AFE95-423A-A081-85F3-AD0D70D0FB4D at 10.44.3.153' in 32000 ms
(Method: INVITE)
itpbx01*CLI>
<--- SIP read from UDP:10.44.3.153:5060 --->
ACK sip:1401 at 10.44.9.0 SIP/2.0
Via: SIP/2.0/UDP
10.44.3.153:5060;rport;branch=z9hG4bK26FAB8569057F005F91285F58781AD32
From: 1501 <sip:1501 at 10.44.9.0>;tag=990832587
To: <sip:1401 at 10.44.9.0>;tag=as4b3729a3
Contact: <sip:1501 at 10.44.3.153:5060>
Call-ID: 448AFE95-423A-A081-85F3-AD0D70D0FB4D at 10.44.3.153
CSeq: 60785 ACK
Max-Forwards: 70
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
itpbx01*CLI>
<--- SIP read from UDP:10.44.14.251:5060 --->


<------------->
itpbx01*CLI>
<--- SIP read from UDP:10.44.3.153:5060 --->

------------------8<---------------------------8<---------------------------8<---------------------------8<---------------------------8<---------


I dont' know what can i do to solve this problem.
First, i obtain 401 Unauthorize (and i don't understand why)....then
404 extension not found.

Someone can help me?
Many thanks!
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