Hi there,
The Problem:
- No SIP devices (Linksys SPA941 phones and a WM6.1 softphone) all on the 192.168.16.0 network can register.
- No SIP Trunks can register (2 Trunks with WorldXchange VXF service, 1 Trunk with 2Talk)
Asterisk Info Reports:
Sip Registry
Host dnsmgr Username Refresh State Reg.Time
2talk.co.nz:5060 N 028nnnnnnnn 120 Unregistered
pan.wxnz.net:5060 N 6825nnnn 120 Request Sent
pan.wxnz.net:5060 N 6825nnnn 120 Request Sent
3 SIP registrations.
Sip Peers
Name/username Host Dyn Nat ACL Port Status
201 (Unspecified) D N A 5060 UNKNOWN
221 (Unspecified) D N A 5060 UNKNOWN
230 (Unspecified) D N A 5060 UNKNOWN
2Talk/028nnnnnnnn 202.180.76.166 N 5060 Unmonitored
6825nnnn/MyUsernamexxxxxx 58.28.20.150 N 5060 Unmonitored
6825nnnn/MyUsername2xxxxx 58.28.20.150 N 5060 Unmonitored
6 sip peers [Monitored: 0 online, 3 offline Unmonitored: 3 online, 0 offline]
The Environment:
I have just done a complete rebuild of my Asterisk box using the
AsteriskNow 1.7.0 iso distro from: http://www.asterisk.org/downloads/asterisknow/i386/asterisknow32.iso).
This installs Centos 5.5 and Asterisk 1.6x.
After install I updated FreePBX to the latest builds:
Asterisk 1.6.2.7
Core 2.7.0.7 FreePBX
DAHDi Config 2.7.0
Feature Code Admin 2.7.0.0
FreePBX Framework 2.7.0.2
System Dashboard 2.7.0.2
Voicemail 2.7.0.1
Info Services 2.7.0.0
Music on Hold 2.7.0.5
Recordings 3.3.9.4
Asterisk Info 2.7.0.0
Backup & Restore 2.7.0.7
Custom Applications 2.7.0.0
The hardware is:
P4 2.8Mhz box with 1Gb RAM
Dual NIC’s
OpenVox card with 4 FXO’s installed (actually no longer needed)
The Network:
eth0:
IP 192.168.16.0
Subnet 255.255.255.0
Static IP
no default Gateway
attached to my fully internal network and this is where my SIP phones live.
eth1:
IP 10.1.1.5
Subnet 255.255.255.0
Configured by DHCP but assigned a fixed address based on the MAC address.
Default GW 10.1.1.1
This is in my “quazi-DMZ” zone talking to a DLink DSL-502T Router.
Also on the network is a Win2003 SBS machine which does among other things routing to the Internet between the the two Zones.
The D-Link Router has Ports 5000-6000 UDP/TCP opened and pointing to IP 10.1.1.5
And Ports 10000-20000 also pointing to IP 10.1.1.5 (the Asterisk box)
What have I tried:
Following some of the posts I have the following in /etc/asterisk/sip_general_custom.conf:
registersip=yes
nat=yes
externhost=10.1.1.5
localnet=192.168.16.0/255.255.255.0
externrefresh=10
Internet and routing appears to be working correctly:
Internal network -
[root@PBX01 ~]# traceroute 192.168.16.10
traceroute to 192.168.16.10 (192.168.16.10), 30 hops max, 40 byte packets
1 pbx01.avs.local (192.168.16.200) 3000.994 ms !H 3000.980 ms !H 3000.965 ms !H
Internet addresses:
[root@PBX01 ~]# traceroute 58.28.20.150
traceroute to 58.28.20.150 (58.28.20.150), 30 hops max, 40 byte packets
1 mygateway1.ar7 (10.1.1.1) 0.800 ms 1.286 ms 1.553 ms
2 ip-58-28-15-31.wxnz.net (58.28.15.31) 95.126 ms 96.276 ms 98.468 ms
3 ip-58-28-13-83.wxnz.net (58.28.13.83) 54.891 ms 55.868 ms 56.734 ms
4 ge-0-0-0-akl-core3.wxnz.net (58.28.8.7) 57.999 ms 58.882 ms 60.767 ms
5 * * *
6 * * *
7 * * *
8 * * *
9 * * *
SELinux and IPTables are both disabled.
When I tried checking for ports however I couldn’t connect on 5060 (though this may also be my own ineptitude as my Linux knowledge is rusty and limited).
VXF Trunk settings as entered through FreePBX:
==============================================================
Trunk Description: AVSVXFLine
Outbound CID: 06825nnnn
CID Options: Any
Max Channels: 2
Trunk Name: 06825nnnn
Peer Details:
useragent=Asterisk PBX
regseconds=180
registertimeout=20
register=6825nnnn:MySecretxxxxxxxxx:[email protected]/6825nnnn
port=5060
language=en
dtmfmode=rfc2833
disallow=all
context=default
bindport=5060
bindaddr=0.0.0.0
allowoverlap=no
allow=g729&ulaw&alaw
nat=yes
User Context: 6825nnnn
User Details:
username=MyUsernamexxxxxxx
type=peer
secret=MySecretxxxxxxxx
nat=yes
insecure=invite,port
host=pan.wxnz.net
fromuser=6825nnnn
fromdomain=pan.wxnz.net
canreinvite=no
Register String:
6825nnnn:MySecretxxxxxxxxxx:[email protected]/6825nnnn
================================================
SIP Extensions:
================================================
This device uses sip technology.
secret MySecret
dtmfmode rfc2833
canreinvite no
context from-internal
host dynamic
type friend
nat yes
port 5060
qualify yes
callgroup
pickupgroup
disallow
allow
dial SIP/201
accountcode
mailbox 201@default
deny 0.0.0.0/0.0.0.0
permit 0.0.0.0/0.0.0.0
==================================================
I have istalled Asterisk a number of times without having trouble in the past. The difference this time is using the dual NIC scenario as I wanted to delete the PAPT devices that were previously in the DMZ zone and were used to connect to WorldxChange which was at the time my only VoIP provider.
What suggestions / tests / configuration changes can you offer? Please be specific with your instructions, since although I know my way around a keyboard my Linux is limited.
Thanks a heap,
Andrew