Asterisk New-bee here. Trying to setup a test server at home 3

[reelbigfish]

Hello all - would someone elaborate a little bit on how an Asterisk bigginer would setup a test server at home.

will it work using VMware/CentOS on a xp PC?

I really don't know where to begin. Any help will be greatly appreciated. Thanks

 

[busster]

Yes, it will work, but only really recommended for a lab or testing type of environment.
I would suggest that you go to Trixbox.org and download their ISO. This installs Centos, Asterisk, FreePBX GUI, and mysql. It has been my experience that people who download Asterisk end up adding a GUI, usually FreePBX, so it ends up looking like Trixbox anyway.
The forums on Trixbox.org are great and they have many people there willing to help. Alternate choice would be AstNow, but they just do not have the forum support group that TrixBox does.
Bottom line, an older PC with 512 memory or better 1G, boot up to the CD and within 45 min or so, you can have an operating system.

 

[engjohn]

Yes you can run it in a VM, although you will most likely have timing issues which will present as jittery audio.

But for testing, go for it. I also recommend Elastix, Trixbox, and AsteriskNow...

Have fun.

 

[reelbigfish]

Thanks

Any recommendations for free SIP soft phones?

For testing of course...

 

[AnthonyHardy]

I use X-lite here in our offices. It seems to work well, though we haven't used it as a "production" phone, just for testing, like you mention.

It's made by CounterPath:

http://www.counterpath.com/

 

[busster]

Agree with AnthonyHardy, x-lite is good. Free version has some limits, but works.

Think there is also a free one on 3cx.com.

 

[reelbigfish]

Cool - Quick question:

How do you turn off the firewall in the Asterisk Box?

I have the server up but can't get the phone to register. I think its a firewall issue on the Asterisk Box.

 

[engjohn]

I do not believe that there is a firewall setup by default. I have never had that happen...

 

[AnthonyHardy]

reelbigfish,

can you get to the internet from the asterisk box? perhaps just pinging the phone? Can you ping the phone from your xp pc? Making sure the two can talk using ping is your first goal.

Some vm's have networking issues and need a little tweaking on the vm setup side (not the OS side)

 

[reelbigfish]

AnthonyHardy - Yes, I can Ping the box from the SIP Softphone agent/client station and I can also access the internet via the Asterisk box. I was able to download serval of the needed components for Asterisk.

I don't think the Softphone is registering to asterisk at all.

0S = CentOS

The SIP soft phone is on a XP WS on the same LAN as the Asterisk box.

Also - How can I view real time call traces in Asterisk? This might help me troubleshoot.

I have a WireShark Sniff of my latest attempt with the soft phone.

 

[engjohn]

from the console of the asterisk box, type "asterisk -vvvvvvr"
That will start an asterisk console with verbosity set to 6
That will show you in realtime what is happening...

 

[reelbigfish]

engjohn - thank you! How can I capture and save these traces to a doc? I'd like to email them...

 

[engjohn]

To be honest, I have never tried to capture for the console.
I always ssh in from another box (my workstation). I can then just copy and paste from the terminal to an email...

 

[reelbigfish]

how do you run the ssh? via virtual termail? anything special?

 

[engjohn]

From a Linux or Mac system just open the terminal and type
ssh <ip.of.your.server>

On windows, download and run putty

 

[reelbigfish]

Cool - I'll look for "putty" Thanks

 

[reelbigfish]

I tried Putty and It worked great Thanks engjohn!

 

[reelbigfish]

Success! I managed to get two Cisco 7960 SIP Phones and an X-Lite soft-phone registered to my Asterisk server and were all able to call each other back and forth. It was a bit rocky on the first few runs due to a couple syntax/spelling errors in my part. You really have to be careful and sometimes a little creative when reading different material.

Now I'll work on getting connected to a PSTN. I'll keep this thread posted. Thanks

Here is my activity log:

FYI - the NOTICE stuff toward the end turned out to be the X-Lites V-mail setting, check for V-mail was checked ON.


localhost*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
1002/1002 192.168.1.107 D 5060 Unmonitored
1001/1001 192.168.1.105 D 5060 Unmonitored
1000/1000 192.168.1.100 D 45984 Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]
[Aug 10 20:06:35] NOTICE[2829]: chan_sip.c:15935 handle_request_subscribe: Received SIP subs cribe for peer without mailbox: 1000
-- Executing [1000@phones:1] Verbose("SIP/1002-082cb1d8", "1|Extension 1000") in new stack
Extension 1000
-- Executing [1000@phones:2] Dial("SIP/1002-082cb1d8", "SIP/1000|30") in new stack
-- Called 1000
-- SIP/1000-082d0cd8 is ringing
-- SIP/1000-082d0cd8 answered SIP/1002-082cb1d8
-- Native bridging SIP/1002-082cb1d8 and SIP/1000-082d0cd8
== Spawn extension (phones, 1000, 2) exited non-zero on 'SIP/1002-082cb1d8'
-- Executing [1000@phones:1] Verbose("SIP/1001-082cb1d8", "1|Extension 1000") in new stack
Extension 1000
-- Executing [1000@phones:2] Dial("SIP/1001-082cb1d8", "SIP/1000|30") in new stack
-- Called 1000
-- SIP/1000-082d0cd8 is ringing
-- SIP/1000-082d0cd8 answered SIP/1001-082cb1d8
-- Native bridging SIP/1001-082cb1d8 and SIP/1000-082d0cd8
== Spawn extension (phones, 1000, 2) exited non-zero on 'SIP/1001-082cb1d8'
-- Executing [1001@phones:1] Verbose("SIP/1000-082cb1d8", "1|Extension 1001") in new stack
Extension 1001
-- Executing [1001@phones:2] Dial("SIP/1000-082cb1d8", "SIP/1001|30") in new stack
-- Called 1001
-- SIP/1001-082d0cd8 is ringing
-- SIP/1001-082d0cd8 answered SIP/1000-082cb1d8
-- Native bridging SIP/1000-082cb1d8 and SIP/1001-082d0cd8
== Spawn extension (phones, 1001, 2) exited non-zero on 'SIP/1000-082cb1d8'
-- Executing [1001@phones:1] Verbose("SIP/1000-082cb1d8", "1|Extension 1001") in new stack
Extension 1001
-- Executing [1001@phones:2] Dial("SIP/1000-082cb1d8", "SIP/1001|30") in new stack
-- Called 1001
-- SIP/1001-082d0cd8 is ringing
-- SIP/1001-082d0cd8 answered SIP/1000-082cb1d8
-- Native bridging SIP/1000-082cb1d8 and SIP/1001-082d0cd8
== Spawn extension (phones, 1001, 2) exited non-zero on 'SIP/1000-082cb1d8'
-- Executing [1002@phones:1] Verbose("SIP/1000-082cb1d8", "1|Extension 1002") in new stack
Extension 1002
-- Executing [1002@phones:2] Dial("SIP/1000-082cb1d8", "SIP/1002|30") in new stack
-- Called 1002
-- SIP/1002-082d0cd8 is ringing
-- SIP/1002-082d0cd8 answered SIP/1000-082cb1d8
-- Native bridging SIP/1000-082cb1d8 and SIP/1002-082d0cd8
[Aug 10 20:09:35] NOTICE[2829]: chan_sip.c:15935 handle_request_subscribe: Received SIP subscribe fo r peer without mailbox: 1000
== Spawn extension (phones, 1002, 2) exited non-zero on 'SIP/1000-082cb1d8'
-- Executing [1001@phones:1] Verbose("SIP/1002-082cb1d8", "1|Extension 1001") in new stack
Extension 1001
-- Executing [1001@phones:2] Dial("SIP/1002-082cb1d8", "SIP/1001|30") in new stack
-- Called 1001
-- SIP/1001-082d0cd8 is ringing
-- SIP/1001-082d0cd8 answered SIP/1002-082cb1d8
-- Native bridging SIP/1002-082cb1d8 and SIP/1001-082d0cd8
== Spawn extension (phones, 1001, 2) exited non-zero on 'SIP/1002-082cb1d8'
-- Unregistered SIP '1000'
-- Registered SIP '1000' at 192.168.1.100 port 28028
localhost*CLI>