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Martin Hope
fileinsert
Asked: 2017-01-28 05:08:50 +0800 CST
  • 1

I have some clients connected to my Asterisk server behind a NAT device. When using TCP everything works OK. If I change the client and server config to use UDP (from transport=tcp to transport=udp,tcp or even simply transport=udp) the phone can no longer register and Asterisk sends SIP: SIP/2.0 401 Unauthorized to the client.

From what I have read it is possible to use UDP behind a NAT with Asterisk, however experience seems to dictate otherwise. Have I missed another mandatory step or is there something else that I am unaware of?

To clarify: it is the clients behind the NAT device; not the server.

Here's the relevant portions of my sip.conf:

[general]
context=bogus
allowguest=no
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
disallow=all
allow=ulaw
alwaysauthreject=yes
videosupport=yes
canreinvite=no
nat=force_rport,comedia
session-timers=refuse
localnet=x.x.x.x/255.255.0.0     ; blanked out for serverfault
tcpenable=yes                    ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0:5060         ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
transport=udp,tcp

[my-codecs](!)                    ; a template for my preferred codecs
        disallow=all
;        allow=ilbc
;        allow=gsm
;        allow=g723
        allow=alaw
        allow=ulaw
;       allow=g722

[basic-options](!,my-codecs)                ; another template
        dtmfmode=rfc2833
        context=internal
        type=friend


[natted-phone](!,basic-options)   ; another template inheriting basic-options
        directmedia=no
        host=dynamic
        transport=tcp,udp

[101](natted-phone)
callerid="Office"
secret=12345678901234567890
context=internal
mailbox=101@main
transport=udp,tcp

UPDATE: On investigation I have found that this only seems to affect Avaya handsets. Other softphones connected to the same server do not seem to suffer this problem. Here is a sample of register and un authorized packets:

asterisk*CLI> sip set debug on
SIP Debugging enabled
<--- SIP read from UDP:x.x.x.x:1032 --->
REGISTER sip:asterisk SIP/2.0
From: sip:2003@asterisk;tag=46c2abf4589053ec5895ffc0_F2003192.168.0.174
To: sip:2003@asterisk
Call-ID: [email protected]
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 192.168.0.174;branch=z9hG4bK1_5a7d12149bcad5895ffc4_R2003
Content-Length: 0
Max-Forwards: 70
Contact: <sip:[email protected];avaya-sc-enabled;transport=udp>;q=1;expires=900;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-2cf4c54ef19b>"
Allow: INVITE,CANCEL,BYE,ACK,SUBSCRIBE,NOTIFY,MESSAGE,INFO,PUBLISH,REFER,UPDATE
User-Agent: Avaya one-X Deskphone
Supported: eventlist

<------------->
--- (12 headers 0 lines) ---
Sending to x.x.x.x:1032 (NAT)
Sending to x.x.x.x:1032 (NAT)

<--- Transmitting (NAT) to x.x.x.x:1032 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.174;branch=z9hG4bK1_5a7d12149bcad5895ffc4_R2003;received=x.x.x.x;rport=1032
From: sip:2003@asterisk;tag=46c2abf4589053ec5895ffc0_F2003192.168.0.174
To: sip:2003@asterisk;tag=as5f37e30b
Call-ID: [email protected]
CSeq: 1 REGISTER
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2350c222"
Content-Length: 0
udp nat asterisk sip avaya
Martin Hope
Rick
Asked: 2012-07-19 06:53:15 +0800 CST
  • 3

I looked at many articles and can;t quite seem to find an answer that work so here goes. I'm setting up a DHCP server running Ubuntu 12.04 LTS, but I'm unable to get the phones to get an IP address. These are Avaya phones that require the Option-176 with all of the string that follows see below. If I comment out the DHCP information for one of the VLANs and connect the phone is seems to work fine but on the actual vlan for phones no go. I'm sure the fix is simple and hope that you can point it out for me. Thanks.

From the dhcpd.conf

shared-network OVKData {
subnet 192.168.226.0 netmask 255.255.255.0 {
    option routers 192.168.226.1;
    option broadcast-address 192.168.226.255;
    range 192.168.226.30 192.168.226.100;
    option option-176 "MCIPADD=192.168.30.3,192.168.30.4,MCPORT=1719,TFTPSRVR=192.168.222.129,L2Q=1,L2QVLAN=888,VLANTEST=0";
#   option option-176 "L2Q=1,L2QVLAN=888,VLANTEST=0";
#   option option-176 "MCIPADD=192.168.30.3,192.168.30.4,MCPORT=1719,TFTPSRVR=192.168.222.129";
    }
#shared-network OVKVoice {
subnet 192.168.98.0 netmask 255.255.255.0 {
    option routers 192.168.98.1;
    option broadcast-address 192.168.98.255;
    range 192.168.98.30 192.168.98.100;
#   option option-176 "MCIPADD=192.168.30.3,192.168.30.4,MCPORT=1719,TFTPSRVR=192.168.222.129,L2Q=1,L2QVLAN=888,VLANTEST=0";
    option option-176 "L2Q=1,L2QVLAN=888,VLANTEST=0";
    option option-176 "MCIPADD=192.168.30.3,192.168.30.4,MCPORT=1719,TFTPSRVR=192.168.222.129";
    }
}

From the interfaces

auto vlan226
auto vlan888
#auto vlan300

# OVKData vlan888
iface vlan226 inet static
    address 192.168.226.4
    netmask 255.255.255.0
    gateway 192.168.226.1
    network 192.168.226.0
    broadcast 192.168.226.255
    mtu 1500
    vlan_raw_device eth0

# OVKVoice Vlan 888
iface vlan888 inet static
    address 192.168.98.4
    netmask 255.255.255.0
    gateway 192.168.98.1
    network 192.168.98.0
    broadcast 192.168.98.255
    mtu 1500
    vlan_raw_device eth0

FRom Syslog

Jul 17 13:20:17 NOCDHP dhcpd: DHCPOFFER on 192.168.226.32 to 00:04:0d:29:3b:f9 via vlan888
Jul 17 13:20:17 NOCDHP dhcpd: DHCPDISCOVER from 00:04:0d:29:3b:f9 via vlan888
Jul 17 13:20:17 NOCDHP dhcpd: DHCPOFFER on 192.168.226.32 to 00:04:0d:29:3b:f9 via vlan888
Jul 17 13:20:17 NOCDHP dhcpd: DHCPDISCOVER from 00:04:0d:29:3b:f9 via 192.168.98.5
Jul 17 13:20:17 NOCDHP dhcpd: DHCPOFFER on 192.168.226.32 to 00:04:0d:29:3b:f9 via 192.168.98.5
Jul 17 13:20:17 NOCDHP dhcpd: DHCPDISCOVER from 00:04:0d:29:3b:f9 via 192.168.98.1
Jul 17 13:20:17 NOCDHP dhcpd: DHCPOFFER on 192.168.226.32 to 00:04:0d:29:3b:f9 via 192.168.98.1
Jul 17 13:20:17 NOCDHP dhcpd: DHCPDISCOVER from 00:04:0d:29:3b:f9 via 192.168.98.1
Jul 17 13:20:17 NOCDHP dhcpd: DHCPOFFER on 192.168.226.32 to 00:04:0d:29:3b:f9 via 192.168.98.1
Jul 17 13:20:17 NOCDHP dhcpd: DHCPDISCOVER from 00:04:0d:29:3b:f9 via vlan888
ubuntu dhcp avaya
Martin Hope
SlowNetwork
Asked: 2012-07-11 07:52:03 +0800 CST
  • 1

Client : Windows 7 x32 | Dell Latitude E6510, transferring to another identical machine over the network.. 120mbs.

Transferring to any server (2003 server, 2008, 2008R2, or VM's) I can not get more than 30MBS. The server is plugged into the same switch and is a physical server. I have jumbo frames and flow control enabled. I do not understand why I am not getting GB speeds but I do transferring to another workstation.

Running the LAN Speed test software from client to the server it lags..

It is taking 39 seconds to write 500MB..

What else can I check? I have tried transferring to our new Dell R610 server with similar results. The switches we use are Avaya 5050TD.

I updated drivers, forced the port to 1000mb full duplex, disabled auto tuning of the NIC. Same issue..

I am plugged into the same switch as the server and see these slow speeds. I can transfer from Server to Server with fast speeds (120mbs or so). Speeds from client to server will not go over 30mbs.

windows-server-2008 network-share filesystems avaya
Martin Hope
Yash
Asked: 2011-11-08 07:46:34 +0800 CST
  • 1

I am a complete beginner concercing Microsoft Lync and PBX systems, so please bear with me.

I have set up a Lync standard server which is running fine. We can chat/audio & video conference on our internal network. Now, what I need is to connect the Lync server to our telephony system, but here is where I get totally lost.

We have an Avaya Integral 5 D pbx box and a netConnect AMP 6 panel.

Are these piece of hardware compatible with what I am trying to achieve? If so, what are the steps to follow from here?

Thank you.

pbx telephony lync avaya
Martin Hope
Ted Spence
Asked: 2011-05-19 15:55:49 +0800 CST
  • 5

Our company uses an Avaya IPOffice system. We're relatively light on our call volume, but I'd really like to know where our peaks and valleys are. Is it possible to track call usage on the Avaya IPOffice unit via SNMP?

Here's what I know:

  • Avaya publishes their SNMP libraries on the admin CD (and on the web - http://www.oidview.com/mibs/6889/md-6889-1.html). I've looked through them but can't find what I want.
  • Avaya does have a program called "System Status" that shows all the values I care about, but I wish I could track it through my main SNMP system.

What I would really like to know is one or more of the following:

  • Of my 24 available lines, how many are in use at the moment?
  • How many internal-to-internal conversations are happening at the moment? These do not use our available external lines, but would be interesting to track.
  • How many internal phones are currently off-hook?
monitoring snmp avaya