Help with a basic setup.

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Help with a basic setup.

Postby superbawlz » Fri Apr 06, 2007 9:29 am

Hello, I have been dabling with Asterisk on and off over the last few years. We it came time to set up an actually working PBX with it I discovered AsteriskNow. I downloaded it and realized this a pretty slick little distro. I went through and set up my companies information in the web-based interface. It's listed below. The problem I am running into is that it doesn't answer the incoming call on the FXO card and I can't seem to get the SIP SoftPhone (X-Lite or IDEFISK) to register with the server. Config information is below.

-------------GENERAL--------------------
Currently I have,
(1) FXO card in the PBX with an analog line attached to it.
(6) SIP Extensions (SoftPhones)
(1) Call Queue that should route calls to x201 and x206.
The phone number is (269)649-5622.
The objective is that when a call comes in on (269)649-5622 the PBX will pick it up add it to the queue. Anything in the queue will ring to x201 and x206. From there they can pass it to any of the other extensions.

----------NETWORK CONFIGS-----------
Server-IP: 192.168.1.40
Server-SN: 255.255.255.0
Server-GW: 192.168.1.9
Server-DNS1: 192.168.1.9
-
Workstation-IP: 192.168.1.16
Workstation-SN: 255.255.255.0
Workstation-GW: 192.168.1.9
Workstation-DNS1: 192.168.1.9
*Both are operating via full-duplex 10/100/1000 Ethernet NICs connected by CAT-6 through a Linksys RVS-4000 gigabit router.

--------------PBX CONFIGS--------------
I have created screenshots of my web-based interface menus.
http://www.brigsen.com/asterisk

I also had these under system information.

###GENERAL###

OS Version:
Linux localhost.localdomain 2.6.17.11-1.1.x86.i686.cmov #1 Tue Sep 5 23:54:38 EDT 2006 i686 i686 i386 GNU/Linux

Uptime:
10:23:00 up 23:55, 2 users, load average: 0.00, 0.00, 0.00

Asterisk Build:
Asterisk 1.4.0
Asterisk GUI-version Revision: 355 $

Server Date & TimeZone:
Fri Apr 6 10:23:00 EDT 2007

Hostname:
localhost.localdomain

###IFCONFIG###
eth0 Link encap:Ethernet HWaddr 00:11:11:71:C9:5D
inet addr:192.168.1.40 Bcast:192.168.1.255 Mask:255.255.255.0
inet6 addr: fe80::211:11ff:fe71:c95d/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:196765 errors:0 dropped:0 overruns:0 frame:0
TX packets:78418 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:100845587 (96.1 Mb) TX bytes:41682794 (39.7 Mb)

lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:472569 errors:0 dropped:0 overruns:0 frame:0
TX packets:472569 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:81857582 (78.0 Mb) TX bytes:81857582 (78.0 Mb)

###RESOURCES###
Disk Usage:

Filesystem Size Used Avail Use% Mounted on
/dev/hda2 36G 1.1G 33G 4% /
/dev/hda1 99M 11M 84M 12% /boot
/dev/shm 379M 0 379M 0% /dev/shm

Memory Usage:

total used free shared buffers cached
Mem: 774484 258572 515912 0 37320 94828
-/+ buffers/cache: 126424 648060
Swap: 1044216 0 1044216


###LOGS###
I posted it here because it was too long.
http://www.brigsen.com/asterisk/asterisklog.txt
superbawlz
Newsterisk
 
Posts: 7
Joined: Fri Apr 06, 2007 7:50 am

Postby santosam » Fri Apr 06, 2007 10:24 am

There are quite a few posts about queues problems on AsteriskNow. If you don't use queues can you receive calls?
santosam
Oldsterisk
 
Posts: 137
Joined: Fri Feb 16, 2007 1:47 pm
Location: Portugal

Postby superbawlz » Fri Apr 06, 2007 12:41 pm

OK, I removed the call queue and tried it. It picked the call up this time but it didn't ring through to the SIP Softphone. It went directly to voice mail. This leads me to my earlier thought that for some reason the client isn't registering correctly. I have the client set up properly. I have tested it with other servers I have and third-party servers as well. I think there is something messed up in the PBX configs for it. I'm not sure the right place to even look.

I have provide a few of the config files that I believe might be related.
http://www.brigsen.com/asterisk/configs/

I'm not sure if they are the problem or not though.

Also, I'm getting a "Login Timed Out! Contact Network Admin." on the X-Lite softphone.

-----X-LITE CONFIG------
Enabled: Yes
Display Name: Jackie Hovious
Username: 201
Authorization User: 201
Password: 1234
Domain/Realm: localhost.localdomain
SIP Proxy: 192.168.1.40
Out Bound Proxy: 192.168.1.40
Use Outboand Proxy: default
Send Internal IP: Default
Register: default


###LOG FROM TEST CALL###
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.16 : 5060 (NAT)

<--- Transmitting (no NAT) to 5.91.14.248:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 5.91.14.248:5060;branch=z9hG4bK6188740D75BB4A568B72549572D8C768;received=192.168.1.16;rport=5060
From: Jackie Hovious <sip:201@localhost.localdomain>;tag=1801787333
To: Jackie Hovious <sip:201@localhost.localdomain>
Call-ID: A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain
CSeq: 51170 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:201@192.168.1.40>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 5.91.14.248:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 5.91.14.248:5060;branch=z9hG4bK6188740D75BB4A568B72549572D8C768;received=192.168.1.16;rport=5060
From: Jackie Hovious <sip:201@localhost.localdomain>;tag=1801787333
To: Jackie Hovious <sip:201@localhost.localdomain>;tag=as2e9d7e21
Call-ID: A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain
CSeq: 51170 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a88e17e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '632A76DA71494A859DBA50F4317BB49E@hovaire.net' Method: REGISTER
[Apr 6 14:37:00] NOTICE[2939]: callerid.c:613 callerid_feed: Caller*ID failed checksum
[Apr 6 14:37:02] NOTICE[2939]: chan_zap.c:6357 ss_thread: Got event 18 (Ring Begin)...
Really destroying SIP dialog '552472627b7200f502a9c91938be1452@127.0.0.1' Method: INVITE
[Apr 6 14:37:02] WARNING[2939]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Apr 6 14:37:02] WARNING[2939]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination)

<--- SIP read from 192.168.1.16:5060 --->
REGISTER sip:localhost.localdomain SIP/2.0
Via: SIP/2.0/UDP 5.91.14.248:5060;rport;branch=z9hG4bKA097D804925B4631ABEEF228AB4DCB23
From: Jackie Hovious <sip:201@localhost.localdomain>;tag=1801787333
To: Jackie Hovious <sip:201@localhost.localdomain>
Contact: "Jackie Hovious" <sip:201@5.91.14.248:5060>
Call-ID: A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain
CSeq: 51171 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103a
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.16 : 5060 (NAT)

<--- Transmitting (no NAT) to 5.91.14.248:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 5.91.14.248:5060;branch=z9hG4bKA097D804925B4631ABEEF228AB4DCB23;received=192.168.1.16;rport=5060
From: Jackie Hovious <sip:201@localhost.localdomain>;tag=1801787333
To: Jackie Hovious <sip:201@localhost.localdomain>
Call-ID: A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain
CSeq: 51171 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:201@192.168.1.40>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 5.91.14.248:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 5.91.14.248:5060;branch=z9hG4bKA097D804925B4631ABEEF228AB4DCB23;received=192.168.1.16;rport=5060
From: Jackie Hovious <sip:201@localhost.localdomain>;tag=1801787333
To: Jackie Hovious <sip:201@localhost.localdomain>;tag=as2e9d7e21
Call-ID: A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain
CSeq: 51171 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e25f83e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain' in 32000 ms (Method: REGISTER)

<--- SIP read from 192.168.1.16:5060 --->
REGISTER sip:localhost.localdomain SIP/2.0
Via: SIP/2.0/UDP 5.91.14.248:5060;rport;branch=z9hG4bKA097D804925B4631ABEEF228AB4DCB23
From: Jackie Hovious <sip:201@localhost.localdomain>;tag=1801787333
To: Jackie Hovious <sip:201@localhost.localdomain>
Contact: "Jackie Hovious" <sip:201@5.91.14.248:5060>
Call-ID: A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain
CSeq: 51171 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103a
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.16 : 5060 (NAT)

<--- Transmitting (no NAT) to 5.91.14.248:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 5.91.14.248:5060;branch=z9hG4bKA097D804925B4631ABEEF228AB4DCB23;received=192.168.1.16;rport=5060
From: Jackie Hovious <sip:201@localhost.localdomain>;tag=1801787333
To: Jackie Hovious <sip:201@localhost.localdomain>
Call-ID: A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain
CSeq: 51171 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:201@192.168.1.40>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 5.91.14.248:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 5.91.14.248:5060;branch=z9hG4bKA097D804925B4631ABEEF228AB4DCB23;received=192.168.1.16;rport=5060
From: Jackie Hovious <sip:201@localhost.localdomain>;tag=1801787333
To: Jackie Hovious <sip:201@localhost.localdomain>;tag=as2e9d7e21
Call-ID: A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain
CSeq: 51171 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e25f83e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain' in 32000 ms (Method: REGISTER)

<--- SIP read from 192.168.1.16:5060 --->
REGISTER sip:localhost.localdomain SIP/2.0
Via: SIP/2.0/UDP 5.91.14.248:5060;rport;branch=z9hG4bKA097D804925B4631ABEEF228AB4DCB23
From: Jackie Hovious <sip:201@localhost.localdomain>;tag=1801787333
To: Jackie Hovious <sip:201@localhost.localdomain>
Contact: "Jackie Hovious" <sip:201@5.91.14.248:5060>
Call-ID: A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain
CSeq: 51171 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103a
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.16 : 5060 (NAT)

<--- Transmitting (no NAT) to 5.91.14.248:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 5.91.14.248:5060;branch=z9hG4bKA097D804925B4631ABEEF228AB4DCB23;received=192.168.1.16;rport=5060
From: Jackie Hovious <sip:201@localhost.localdomain>;tag=1801787333
To: Jackie Hovious <sip:201@localhost.localdomain>
Call-ID: A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain
CSeq: 51171 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:201@192.168.1.40>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 5.91.14.248:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 5.91.14.248:5060;branch=z9hG4bKA097D804925B4631ABEEF228AB4DCB23;received=192.168.1.16;rport=5060
From: Jackie Hovious <sip:201@localhost.localdomain>;tag=1801787333
To: Jackie Hovious <sip:201@localhost.localdomain>;tag=as2e9d7e21
Call-ID: A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain
CSeq: 51171 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e25f83e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'A6A38663FCC7495DB695F5C087CF148C@localhost.localdomain' in 32000 ms (Method: REGISTER)
Last edited by superbawlz on Fri Apr 06, 2007 12:50 pm, edited 1 time in total.
superbawlz
Newsterisk
 
Posts: 7
Joined: Fri Apr 06, 2007 7:50 am

Postby santosam » Fri Apr 06, 2007 12:46 pm

On X-Lite you have localhost.localdomain as the Domain/Realm but you must put there the hostname of the asterisk server or the IP 192.168.1.40.
santosam
Oldsterisk
 
Posts: 137
Joined: Fri Feb 16, 2007 1:47 pm
Location: Portugal

Postby superbawlz » Fri Apr 06, 2007 12:55 pm

I have a hosts file redirect localhost.localdomain -> 192.168.1.40. I will try it with the IP though.

I also added to the previous post. That I am getting a "Login Timed Out! Contact Network Admin" error on X-Lite.

Update:

I changed it to the IP Address and now I have,

"Login Failed! Contact Network Admin"

------Log from PBX------


<--- SIP read from 192.168.1.16:5060 --->
REGISTER sip:192.168.1.40 SIP/2.0
Via: SIP/2.0/UDP 5.91.14.248:5060;rport;branch=z9hG4bK31141B605DDA494C9D5E52AC58A773B5
From: Jackie Hovious <sip:201@192.168.1.40>;tag=3310639615
To: Jackie Hovious <sip:201@192.168.1.40>
Contact: "Jackie Hovious" <sip:201@5.91.14.248:5060>
Call-ID: 0F982407F3574289AE77C67E67DE4A0E@192.168.1.40
CSeq: 63894 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103a
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.16 : 5060 (NAT)

<--- Transmitting (NAT) to 192.168.1.16:5060 --->
SIP/2.0 404 Not found (unknown domain)
Via: SIP/2.0/UDP 5.91.14.248:5060;branch=z9hG4bK31141B605DDA494C9D5E52AC58A773B5;received=192.168.1.16;rport=5060
From: Jackie Hovious <sip:201@192.168.1.40>;tag=3310639615
To: Jackie Hovious <sip:201@192.168.1.40>;tag=as5d2b2c19
Call-ID: 0F982407F3574289AE77C67E67DE4A0E@192.168.1.40
CSeq: 63894 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
[Apr 6 14:56:50] NOTICE[2923]: chan_sip.c:14354 handle_request_register: Registration from 'Jackie Hovious <sip:201@192.168.1.40>' failed for '192.168.1.16' - Not a local domain
Scheduling destruction of SIP dialog '0F982407F3574289AE77C67E67DE4A0E@192.168.1.40' in 32000 ms (Method: REGISTER)




----------------------------------------------

P.S. If you have a little bit of time, a messenger, and PayPal I would be willing to send you a little bit of money for your time. Don't have much to work with right this second. We could go $40 or $50 USD to start and $40 - $50 upon fix, $100 USD total. Please don't take it as an insult.
superbawlz
Newsterisk
 
Posts: 7
Joined: Fri Apr 06, 2007 7:50 am

Postby santosam » Fri Apr 06, 2007 1:52 pm

First, and the most important: localhost.localdomain is always 127.0.0.1 - it's basic Unix/Linux. It's configured in /etc/hosts file. What you were getting from the DNS server was that localhost.localdomain is a server with IP 5.91.14.248 and you were trying to REGISTER your softphones on that server. Not an Asterisk problem, just a bad configuration of your PC.....
Second, now you're getting an error from the Asterisk server, which is a good start :-)
The error is that your PC is not a local domain. That means that the domain of your server and the domain of your PC are different...
The Server must be something like asterisk.brigsen.com with the file /etc/hosts with 2 lines:
127.0.0.1 localhost.localdomain localhost
192.168.1.40 asterisk.brigsen.com asterisk
and on your PC a /etc/hosts file similar to the one on top. remember to change the files on your DNS server, if you have a DNS server for you local network.
santosam
Oldsterisk
 
Posts: 137
Joined: Fri Feb 16, 2007 1:47 pm
Location: Portugal

Postby superbawlz » Fri Apr 06, 2007 2:11 pm

Actually on the PC I have defined localhost.localdomain as 192.168.1.40 (The IP of the Asterisk Box). Essentially overriding the default result of localhost -> 127.0.0.1. I will however adjust this to asterisk.brigsen.com.

The 5.x.x.x IP range was caused by Hamachi (a VPN software) which I have since removed. And still have the same result. I will try adjusting the hosts file.

Also, please consider my offer at the end of my last post.
superbawlz
Newsterisk
 
Posts: 7
Joined: Fri Apr 06, 2007 7:50 am

Postby santosam » Mon Apr 09, 2007 2:14 pm

Is it working now?
santosam
Oldsterisk
 
Posts: 137
Joined: Fri Feb 16, 2007 1:47 pm
Location: Portugal


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