Asterisk@Home VoIP Server on IS Business 512k ADSL line

Just got our IS Business ADSL Cisco Router at 17:00 today and installed
a Test Asterisk@Home Server on IP address: 196.211.2.162
for a few days for those folks who would like to test running VoIP across
IS Business ADSL

Asterisk@Home website:http://asteriskathome.sourceforge.net

Phones: Free XLite Softphone from http://www.xten.com/index.php?menu=download

Howto Setup Xlite Softphone to connect to Asterisk
http://asteriskathome.sourceforge.net/handbook/index.html#Section_4.1.1
Display Name: Test User
Username : "Your Ext No"
Authorisation User: "Your Ext No"
Password : 1234
Domain/Realm : 196.211.2.162
SIP Proxy : 196.211.2.162
Outbound Proxy : 196.211.2.162

Sometimes it is best to exit and reload XLite for settings to work.
See the feature list below to test or dial your own extension.
Activate Call Waiting to call your self
*70 Activate Call Waiting (deactivated by default)

Codec: GSM or iLBC
SIP protocol

My Ext is 212
Check Voice via Browser: http://196.211.2.162/cgi-bin/vmail.cgi

Switch Board Panel View of PBX:
http://196.211.2.162/panel/


Please fill in your name nickname next to the extension number
Ext Password: 1234 VoiceMail Password: 1234
220
221
222
223
224
225
226
227
228
229

Feature Code List:-
==============
*411 Directory
*43 Echo Test
*60 Time
*61 Weather (New York weather using Festival to read a text message)
*62 Schedule wakeup call
*65 festival test (your extension is XXX)
*70 Activate Call Waiting (deactivated by default)
*71 Deactivate Call Waiting
*72 Call Forwarding System
*73 Disable Call Forwarding
*77 IVR Recording
*78 Enable Do-Not-Disturb
*79 Disable Do-Not-Disturb
*90 Call Forward on Busy
*91 Disable Call Forward on Busy
*97 Message Center (does no ask for extension)
*98 Enter Message Center
*99 Playback IVR Recording
666 Test Fax
7777 Simulate incoming call

Hope you enjoy and give me a call at Ext 212 otherwise if I am unavailable leave me a voicemail.

Please send me your ping responses and a traceroute to 196.211.2.162

Pinging 196.211.2.162 with 32 bytes of data: From 256k SAIX Diginet to IS ADSL (no traffic thru SAIX-IS peering)
otherwise ping rates double during the day

Reply from 196.211.2.162: bytes=32 time=113ms TTL=53
Reply from 196.211.2.162: bytes=32 time=114ms TTL=53
Reply from 196.211.2.162: bytes=32 time=115ms TTL=53
Reply from 196.211.2.162: bytes=32 time=111ms TTL=53

Home <-> WiFi link <-> 256k SAIX <-> IS ADSL 512 <-> Asterisk PBX

Sorry guys no Zaptel interfaces installed so you cannot make any PSTN
phone calls here in the Western Cape.

I'm sooooo keen to play with this stuff - busy downloading *@H as i type...!!

But with only having a 192k line out - i don't think i'm going to be placing any calls outside my LAN

Look at Amazon or if you can get any books on Asterisk locally
http://www.amazon.com/exec/obidos/s...n&field-keywords=asterisk&Go.x=0&Go.y=0&Go=Go

Excellent source of info
http://www.voip-info.org/tiki-index.php?page=Asterisk at Home

Asterisk is ideal to implement eg Rural community where there is no telkom phones
Town - Telkom PSTN lines = Asterisk PBX > Wireless link <--> IP phones (Prepay) or Softphone or IP hard phone. (shop, farm etc)
Benefits:-
* Free IP based local calls (should upset Telkom)
* Email notification of Voicemail via WAV attachment
* Assign a fix phone number at Asterisk PBX for each IP phone
eg Incoming call on phone no 043 730 1212 -> Ext 212 (farmer brown)
Incoming call on phone no 043 730 1210 -> Ext 213 (spaza shop owner)
Requires Digium ISDN or PSTN cards
* Connect Asterisk PBX to Internet so local community has Email access +++
* Connect now other devices onto Wireless like ie Network IP cams www.axis.com www.mobotix.com, Weather stations

I just been playing around a bit with your asterisk - as expected my line is too slow to have any sort of meaning full conversation. Managed to get the time from your machine though

Ping is a little pathetic

But the number of hops on the trace route is quite promising

good quality VOIP

Got good voice quality after changing codecs to gsm and ilbc.
office notebook PC>lan>ipcop>iBurst>.... remote Asterisk PBX.

I grabbed extension 228. Phone me sometime!

Please help an idiot: I did ping with command prompt.
How do you copy/paste it? ^C and ^V just laugh at me.
Also traceroute gives a rude message (windows OS).

Thanks for those response times seems like ping times
double during the day to ambo, the line is not used
abit disappointing the traceroute times seem to be the norm
for during the day after hours it drops to 115ms
Web based traceroute
http://196.211.2.162/cgi-bin/traceroute.cgi
http://196.211.2.162/cgi-bin/ping.cgi
212 Kilos
220 Ambo (bad ping response, pity)
221
222
223
224
225
226
227
228 User: 8321
229 User: Blade (good response from SAIX ADSL 512)

Dell Optiplex GX240 512Mb P4 2.4Ghz 80 Gb

Remember Voicemail password:1234
Checkout via Web interface
http://196.211.2.162/cgi-bin/vmail.cgi
Dial *98 Extension No : 220 Password 1234

8321 said:

Got good voice quality after changing codecs to gsm and ilbc.
office notebook PC>lan>ipcop>iBurst>.... remote Asterisk PBX.

I grabbed extension 228. Phone me sometime!

Please help an idiot: I did ping with command prompt.
How do you copy/paste it? ^C and ^V just laugh at me.
Also traceroute gives a rude message (windows OS).

Click to expand...

In the dos box right click and in new window chosse mark and select what you want or use select all , then just press <enter> to get the selection onto the clipboard. Then paste in any other application or web forum entry page.
To paste into dogbox is as per usual, right click and "paste"

Sorry forgot to mention - i've been hovering around on ext 220...
But as kilos just found out - its rather a lost cause trying to call me.
The latency and speed is just soooo bad on my connection

ambo said:

Sorry forgot to mention - i've been hovering around on ext 220...
But as kilos just found out - its rather a lost cause trying to call me.
The latency and speed is just soooo bad on my connection

Click to expand...

Use web based traceroute & ping utilities

http://196.211.2.162/cgi-bin/traceroute.cgi
http://196.211.2.162/cgi-bin/ping.cgi

http://196.211.2.162/cgi-bin/traceroute.cgi?t=165.146.47.237

Tried to ping & traceroute your IP address seriously congested
1 196.211.2.161 (196.211.2.161) 1.322 ms 1.654 ms 1.614 ms
2 196.211.129.13 (196.211.129.13) 73.931 ms 67.021 ms 65.650 ms
3 196.36.80.209 (196.36.80.209) 69.149 ms 67.954 ms 68.378 ms
4 196.26.96.203 (196.26.96.203) 70.160 ms 67.442 ms 63.453 ms
5 196.26.0.250 (196.26.0.250) 69.758 ms 70.458 ms 69.961 ms
6 168.209.86.166 (168.209.86.166) 69.306 ms 68.447 ms 64.491 ms
7 196.25.5.29 (196.25.5.29) 75.244 ms 69.452 ms 65.673 ms
8 rndf-ip-er-2-pos-12-1-0.telkom-ipnet.co.za (196.43.11.230) 68.651 ms 67.804 ms 69.064 ms
9 rndf-ip-esr-1-fe-1-0-4.telkom-ipnet.co.za (196.43.10.6) 70.143 ms 69.497 ms 67.068 ms
10 * * *
11 * *

stats you wanted

Ping Gateway from Kilos IS ADSL account

1 196.211.2.162 (196.211.2.162) 0.099 ms 0.297 ms 0.218 ms

Microsoft Windows XP [Version 5.1.2600]
(C) Copyright 1985-2001 Microsoft Corp.

one step - looks suspect?

ping 196.211.2.162

Pinging 196.211.2.162 with 32 bytes of data:

Reply from 196.211.2.162: bytes=32 time=179ms TTL=54
Reply from 196.211.2.162: bytes=32 time=261ms TTL=54
Reply from 196.211.2.162: bytes=32 time=159ms TTL=54
Reply from 196.211.2.162: bytes=32 time=163ms TTL=54

Ping statistics for 196.211.2.162:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 159ms, Maximum = 261ms, Average = 190ms

>

Busy downloading a 640 Mb took 3.5 hrs, maybe thats why the one route is abit suspect

Just got myself a Cheetah-IS account from Clipse - pings to you are looking MUCH better:


And tracert looks much better as well (we are with the same ISP now ):

Tried to get my voicemail with X-Lite but it seems even with a great ping - 192k is not really sufficient.
Which codec is best for slow lines?

Try ilbc codec ideally G729, XLite does not support it only the Xten Pro does
Those responses are great.
Can you dial some of the Features like time, echo test

Is the test box still available?
Extension 212 off the hook, 220, 229 not available. Could leave a message for mysef.

Yeah still available for another 2 weeks till end of month.
My Ext 212 in and out of the office, so do get connected
sometimes

Maybe a bit off subject, but I found your post and tried out your server and it works GREAT!

I also went and got Asterisk@Home and installed it on a machine but for some reason cannot figure out how to get my router to allow the connections from other SIPs. I can connect on my lan where everything seems to be working fine, I can do the echo test, etc, but my brother can't connect to it from his ADSL connection.

Do you maybe have any pointers for me? Possibly something specific that needs to be configured on the Asterisk server since there is NAT involved?

I've got a 384k ADSL to Telkom, quite a few hops to your server, but this still works great for me and the ping time is also pretty cool

Tracing route to 196.211.2.162 over a maximum of 30 hops

1 <1 ms <1 ms <1 ms 192.168.0.254
2 * * * Request timed out.
3 13 ms 12 ms 13 ms 165.146.192.1
4 13 ms 14 ms 13 ms 196.43.10.14
5 36 ms 36 ms 35 ms 196.43.23.150
6 37 ms 36 ms 35 ms 196.25.5.30
7 36 ms 35 ms 38 ms 168.209.86.165
8 39 ms 40 ms 38 ms 196.26.0.14
9 38 ms 37 ms 37 ms 196.26.96.197
10 * 38 ms 37 ms 196.36.80.213
11 90 ms 98 ms 87 ms 196.211.129.14
12 85 ms 85 ms 86 ms 196.211.2.162

Pinging 196.211.2.162 with 32 bytes of data:

Reply from 196.211.2.162: bytes=32 time=86ms TTL=53
Reply from 196.211.2.162: bytes=32 time=84ms TTL=53
Reply from 196.211.2.162: bytes=32 time=87ms TTL=53
Reply from 196.211.2.162: bytes=32 time=88ms TTL=53

Make sure in under your SIP extension that you do the following:
on the Asteisk Management Portal
Nat = yes
Allow = gsm or ilbc

Set your SIP phone up the same codec and it should work
Use the GSM codec uses about 35 kilobits/sec of b/w (upload & Download)