Hello guyz,
I know this is very common theme and there are many already answered topics, but I can’t solve my problem.
I have installed Asterisk 13.6.0 without DAHDI and libpri.
sip show settings:
Global Settings:
----------------
UDP Bindaddress: 192.168.50.197:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Path support : No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: No
Direct RTP setup: No
User Agent: Asterisk PBX 13.6.0
SDP Session Name: Asterisk PBX 13.6.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externaddr: (null)
Externrefresh: 10
Global Signalling Settings:
---------------------------
Codecs: (ulaw|alaw|gsm|h263)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: public
Record on feature: automon
Record off feature: automon
Force rport: Auto (No)
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language: en
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
sip.conf:
[home-users](!)
context=homelan
type=user
disallow=all
allow=alaw
allow=ulaw
[ixt01](home-users)
authname=ixt01
secret=Abc12345
[ixt02](home-users)
secret=Abc12345
extensions.conf:
[homelan]
exten => 101,1,Dial(SIP/ixt01)
exten => 102,1,Dial(SIP/ixt02)
log:
<--- SIP read from UDP:192.168.50.125:51823 --->
REGISTER sip:192.168.50.197;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.50.125:51823;branch=z9hG4bK-d8754z-b94e84f971806a24-1---d8754z-
Max-Forwards: 70
Contact: <sip:ixt01@192.168.50.125:51823;rinstance=7e4b479938e6dfd1;transport=UDP>
To: <sip:ixt01@192.168.50.197;transport=UDP>
From: <sip:ixt01@192.168.50.197;transport=UDP>;tag=d7487267
Call-ID: ZmFhM2FhODE3YzkxODFkZGQyN2M5NWNmMTkyOGE4NjI.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Allow-Events: presence, kpml
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.50.125:51823 (no NAT)
Sending to 192.168.50.125:51823 (no NAT)
<--- Transmitting (no NAT) to 192.168.50.125:51823 --->
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.50.125:51823;branch=z9hG4bK-d8754z-b94e84f971806a24-1---d8754z-;received=192.168.50.125
From: <sip:ixt01@192.168.50.197;transport=UDP>;tag=d7487267
To: <sip:ixt01@192.168.50.197;transport=UDP>;tag=as6a8631a7
Call-ID: ZmFhM2FhODE3YzkxODFkZGQyN2M5NWNmMTkyOGE4NjI.
CSeq: 1 REGISTER
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
What I’m doing wrong? Or where can I look why it doesn’t find accounts exactly?
PS
This is second try to setup sip.conf first time I received not found too, but there is was a record in “sip show subscriptions” and in log was command SUBSCRIBE