SIP/2.0 404 Not found

Asterisk Asterisk Support
1

I got a spiderstar IP PBX, wich is built on Asterisk 1.6.0.15
When trying to register my 1st softphone (192.168.30.115), I am getting 404 not found error.

user - is a new account i created in spiderstar.

Below is a wireshark UDP trace from the computer running firefly.

REGISTER sip:192.168.30.240 SIP/2.0
To: usersip:user@192.168.30.240
From: usersip:user@192.168.30.240;tag=7a297947
Via: SIP/2.0/UDP 192.168.30.115:5060;branch=z9hG4bK-d87543-341661065-1–d87543-;rport
Call-ID: 1d0abb4c094cb913
CSeq: 1 REGISTER
Contact: sip:user@192.168.30.115:5060;expires=7200
Expires: 7200
Max-Forwards: 70
User-Agent: Firefly
Content-Length: 0

SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.30.115:5060;branch=z9hG4bK-d87543-341661065-1–d87543-;received=192.168.30.115;rport=5060
From: usersip:user@192.168.30.240;tag=7a297947
To: usersip:user@192.168.30.240;tag=as7fc9a87a
Call-ID: 1d0abb4c094cb913
CSeq: 1 REGISTER
User-Agent: Asterisk PBX 1.6.0.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

2

It is not enough. What account on asterisk? Post result of sip show peers.

3

.

4

enter to asterisk console( asterisk -r) then issue sip show peers

5

when I type asterisk -r, ubuntu says it’s not installed.
I thought I saw asterisk in server’s reply ?!

6

No what you saw the user agent line. Anyone can put anything on those lines. It is not a clear indicator of what the software is.

The most likely reason though is, spiderstar is probably taking Asterisk and building a package out of it. The actually running executable has been called something else rather than asterisk. Which is why when you type in asterisk at the command line it doesn’t find it.

Visiting their site at http://www.spiderstar.com/en/, I find on their download server this startup script.
It shows that they have named their service spiderstar.

#!/bin/sh

ROOTDIR="/telefon"

case "$1" in
    start)
		cp -f /etc/hosts /etc/resolv.conf ${ROOTDIR}/etc/ >/dev/null 2>&1
		mount proc -t proc ${ROOTDIR}/proc >/dev/null 2>&1
		chroot ${ROOTDIR} /etc/init.d/spiderstar start
		;;

    stop)
        chroot ${ROOTDIR} /etc/init.d/spiderstar stop
        umount ${ROOTDIR}/proc >/dev/null 2>&1
        ;;

    restart)
        $0 stop
        $0 start
        ;;

    status)
        chroot ${ROOTDIR} /etc/init.d/spiderstar status
        ;;

    *)
        echo "Usage: $0 {start|stop|restart|status}"
        exit 1
        ;;
esac

exit 0
7

still not having any luck.

can you ssh to it?

69.242.20.86 root/baba

8

I quick checked you machine. On it Asterisk 1.6.0.15, it installed with uncommon way for UBUNTU, thats why asterisk -r do not give any results. When you login issue “chroot /telefon” then you can use “asterisk -r”, then you see asterisk console, and then you can use “sip show peers”.
I do not know where should configuration from web interface be, but I have added user “siptst”, with password “test”. I do not know can you see it from Web or not, but I think you should connect to server.
now if you issue “sip show peers”, youll see filliwing:

localhost*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status     Realtime
siptst/siptst              (Unspecified)    D          5060     Unmonitored
1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline]
9

Thank you SterhTG!

I was able to connect using the username/pass you created, but I could not see that user in the web console.

REGISTER sip:192.168.30.240 SIP/2.0
To: siptstsip:siptst@192.168.30.240
From: siptstsip:siptst@192.168.30.240;tag=6c569c23
Via: SIP/2.0/UDP 192.168.30.115:5060;branch=z9hG4bK-d87543-417536061-1–d87543-;rport
Call-ID: 1620d020d50a2332
CSeq: 1 REGISTER
Contact: sip:siptst@192.168.30.115:5060;expires=7200
Expires: 7200
Max-Forwards: 70
User-Agent: Firefly
Content-Length: 0

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.30.115:5060;branch=z9hG4bK-d87543-417536061-1–d87543-;received=192.168.30.115;rport=5060
From: siptstsip:siptst@192.168.30.240;tag=6c569c23
To: siptstsip:siptst@192.168.30.240;tag=as7bf05fbd
Call-ID: 1620d020d50a2332
CSeq: 1 REGISTER
User-Agent: Asterisk PBX 1.6.0.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3c4682ec"
Content-Length: 0

REGISTER sip:192.168.30.240 SIP/2.0
To: siptstsip:siptst@192.168.30.240
From: siptstsip:siptst@192.168.30.240;tag=6c569c23
Via: SIP/2.0/UDP 192.168.30.115:5060;branch=z9hG4bK-d87543-693264272-1–d87543-;rport
Call-ID: 1620d020d50a2332
CSeq: 2 REGISTER
Contact: sip:siptst@192.168.30.115:5060;expires=7200
Expires: 7200
Max-Forwards: 70
User-Agent: Firefly
Authorization: Digest username=“siptst”,realm=“asterisk”,nonce=“3c4682ec”,uri=“sip:192.168.30.240”,response=“8ce8bef2e46775120a869972a69f0635”,algorithm=MD5
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.30.115:5060;branch=z9hG4bK-d87543-693264272-1–d87543-;received=192.168.30.115;rport=5060
From: siptstsip:siptst@192.168.30.240;tag=6c569c23
To: siptstsip:siptst@192.168.30.240;tag=as7bf05fbd
Call-ID: 1620d020d50a2332
CSeq: 2 REGISTER
User-Agent: Asterisk PBX 1.6.0.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 3600
Contact: sip:siptst@192.168.30.115:5060;expires=3600
Date: Sun, 10 Oct 2010 12:48:53 GMT
Content-Length: 0

10

looking at the web gui, like i can get the peers screen

sip users

[code]
Name/username Host Dyn Nat ACL Port Status Realtime
siptst/siptst 192.168.30.115 D 5060 Unmonitored
1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline]

Host Username Refresh State Reg.Time
0 SIP registrations.[/code]