Xl-ite Softphone with Astrisk

Hello There, I have setup my astrisk V:1.6 at the home i have a router Linksys WAG200G with open the rquried ports (Picture 1), I installed X-Lite softphone V: 3.0 Build 53621 in my machine WinXp prof i setup the SIP.conf and extension.conf.
My problem its while i try to regsiter the sip in the x-lite and am tic usig the regsiter with domain and recive incomming call its give me this error (Picture 2), If i didn't tic this option its regsiter but iwhile i try to call the extension its give call failed request time out the i hear the the person your are call not available please try again !! as (Picture 3).

I need help to regsiter my sip using x-liet with tic option and if you can provide me a sample configration of sip and extensio so i ca test that

picture1.JPG
picture2.JPG
picture3.JPG
LVL 1
NemCertAsked:
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feptiasCommented:
"sip show peers" is now showing an IP address for one of your peer devices - 192.168.1.104. What device is that? Where did the IP address come from - did you put that address in the sip.conf file or has the device registered with that address? Is it the X-Lite?

If you see no SIP packets when sip debugging is enabled, then you must look again at the settings on the X-Lite and the configuration of your network. The X-Lite should be sending a REGISTER packet to the address 192.168.1.100 (your domain). Are you sure that is the correct address for the Asterisk server? It will not send the REGISTER packet *all* the time - perhaps only every 15 minutes, maybe even longer. However, if you stop and restart the X-Lite then it should send the request on restart. If you see nothing, then check for things like firewalls and routing.

It should be very straightforward for the softphone to send this request over a LAN. Can you ping the Asterisk server from the PC that is hosting the X-Lite softphone?

Is it possible that your Asterisk server has a software firewall enabled? Most default Linux installations include the iptables firewall. Sometimes you have to disable it or modify the rules for the INPUT chain to allow any source to send UDP packets to port 5060.
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memo_tntCommented:
Hi

click on register with domain as pic2

go to your asterisk CLI

by asterisk -rvv

then hit >show peers
and send results


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NemCertAuthor Commented:
fedora*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status    
1001/1001                  (Unspecified)    D   N      5060     Unmonitored
2002/2002                  (Unspecified)    D   N      5060     Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
fedora*CLI
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memo_tntCommented:
are you trying with your xlite from NATed network ??

are you behind a NAT ??
are you in a separate network than your server??

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NemCertAuthor Commented:
No Am in th same network, Astrisk and two winxp machine (X-Lite)  in the same network
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memo_tntCommented:
then you need to set the extensions profile from the FReePBX GUI beside the NAT = no

or vi /etc/asterisk/sip_additional.conf

and put for each extension
nat=0

save to this file
and restart your asterisk

by
 amportal restart


then try after that


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NemCertAuthor Commented:
I'll try now but i didn;t have GUI, by the code

memo can you please provide me a simple sip.con and extension to useing SIP to be sure the configration files are working fine ?,
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NemCertAuthor Commented:
Same :(  its not regsiter same output
regisrtartion error408 request time out
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memo_tntCommented:
plz post sip show peers again
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NemCertAuthor Commented:
fedora*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status    
1001/1001                  (Unspecified)    D   N      5060     Unmonitored
2002/2002                  (Unspecified)    D   N      5060     Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
fedora*CLI>
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memo_tntCommented:
ok now post content of sip_additional.conf
in
 /etc/asterisk/sip_additional.conf


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NemCertAuthor Commented:
Contains the file is :
nat=0

Please i need to solve the problem please provide me the two conf file for sip and extension so i\ll upoladed to my system and check
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feptiasCommented:
The username on the phone is 807. The username and device name on Asterisk is 1001 and 2002. The names need to match - so does the password.
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NemCertAuthor Commented:
Hello there, I change the sip.conf file with the following parameters (Displayname and username and password and autheintcationname to 1001). Its the same problem.

Please provide me a sip.conf and extension.conf as good sample to be consdier where should be the problem, or if you like i can pust the files here so we can modified togther ..

\Regards,
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feptiasCommented:
If the phone is not registering, then extensions.conf is not relevant - only sip.conf.

The error "Request timeout" is likely to indicate that the phone is sending the wrong IP address to Asterisk - so Asterisk is sending a response but the phone doesn't receive it. Do you have STUN enabled on the X-Lite because that can cause it to send an external instead of an internal IP address?

Please can you go to the Asterisk command line interface (e.g. from Linux prompt, type "asterisk -r") and then enter the following command at the Asterisk CLI:
sip set debug on

With sip debug now switched on, all SIP messages to and from Asterisk will appear on the screen. Wait for  the X-Lite to register again (or possibly force it to attempt to re-register by ticking the box "Register with Domain...") and capture the SIP messages that the phone is sending to Asterisk and post back here. To switch of sip debug the command is "sip set debug off".
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NemCertAuthor Commented:
Si nithing appears :
[root@fedora ~]# asterisk -r
Asterisk 1.6.1.10, Copyright (C) 1999 - 2009 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.1.10 currently running on fedora (pid = 2124)
fedora*CLI> sip set debug on
SIP Debugging re-enabled
fedora*CLI> sip set debug off
SIP Debugging Disabled
fedora*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status    
xxx/xxx                    192.168.1.104               5060     Unmonitored
yyy/yyy                    (Unspecified)    D          5060     Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
fedora*CLI> sip set debug on
SIP Debugging enabled
fedora*CLI>
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NemCertAuthor Commented:
Am not enable the STUN
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memo_tntCommented:
that's good,, but it needs to register
so,, you need to check register with domain and receive incoming calls
in your xlite

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NemCertAuthor Commented:
fepias brother :) There is a good news the phone is register upon your advice the problem it was firwall on the astreisk server it was two thing the GUI firwall was started, Seconds thing and the main  as you mention iptables firewall i  run the following command
# /etc/init.d/iptables save
# /etc/init.d/iptables stop

Output as the following :

--- (13 headers 13 lines) ---
Ignoring this INVITE request
<--- SIP read from UDP://192.168.1.102:8832 --->
ACK sip:7888@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:8832;branch=z9hG4bK-d8754z-7e010934903b6472-1---d8754z-;rport
To: "7888"<sip:7888@192.168.1.100>;tag=as5ea70deb
From: "1001"<sip:1001@192.168.1.100>;tag=2c22503e
Call-ID: YzBkZDc0MzJjZjJhNjUxYTdhM2FmNGNhZjU0MzFlZWE.
CSeq: 2 ACK
Content-Length:
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP://192.168.1.102:8832 --->
ACK sip:7888@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:8832;branch=z9hG4bK-d8754z-7e010934903b6472-1---d8754z-;rport
To: "7888"<sip:7888@192.168.1.100>;tag=as5ea70deb
From: "1001"<sip:1001@192.168.1.100>;tag=2c22503e
Call-ID: YzBkZDc0MzJjZjJhNjUxYTdhM2FmNGNhZjU0MzFlZWE.
CSeq: 2 ACK
Content-Length:
<------------->
--- (7 headers 0 lines) ---


After register the phone i need to do the call test between two extension by SIP protocol, Wait
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NemCertAuthor Commented:
Thanks Mr.Voip ..
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