[Last Call] Learn how to a build a cloud-first strategyRegister Now

x
?
Solved

Xl-ite Softphone with Astrisk

Posted on 2009-12-28
20
Medium Priority
?
735 Views
Last Modified: 2013-11-12
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
0
Comment
Question by:NemCert
  • 11
  • 6
  • 3
20 Comments
 
LVL 16

Expert Comment

by:memo_tnt
ID: 26136092
Hi

click on register with domain as pic2

go to your asterisk CLI

by asterisk -rvv

then hit >show peers
and send results


0
 
LVL 1

Author Comment

by:NemCert
ID: 26141731
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
0
 
LVL 16

Expert Comment

by:memo_tnt
ID: 26143530
are you trying with your xlite from NATed network ??

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

0
Industry Leaders: We Want Your Opinion!

We value your feedback.

Take our survey and automatically be enter to win anyone of the following:
Yeti Cooler, Amazon eGift Card, and Movie eGift Card!

 
LVL 1

Author Comment

by:NemCert
ID: 26146443
No Am in th same network, Astrisk and two winxp machine (X-Lite)  in the same network
0
 
LVL 16

Expert Comment

by:memo_tnt
ID: 26148678
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


0
 
LVL 1

Author Comment

by:NemCert
ID: 26149540
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 ?,
0
 
LVL 1

Author Comment

by:NemCert
ID: 26149661
Same :(  its not regsiter same output
regisrtartion error408 request time out
0
 
LVL 16

Expert Comment

by:memo_tnt
ID: 26152065
plz post sip show peers again
0
 
LVL 1

Author Comment

by:NemCert
ID: 26158776
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>
0
 
LVL 16

Expert Comment

by:memo_tnt
ID: 26158908
ok now post content of sip_additional.conf
in
 /etc/asterisk/sip_additional.conf


0
 
LVL 1

Author Comment

by:NemCert
ID: 26159791
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
0
 
LVL 19

Expert Comment

by:feptias
ID: 26161926
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.
0
 
LVL 1

Author Comment

by:NemCert
ID: 26162456
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,
0
 
LVL 19

Expert Comment

by:feptias
ID: 26162829
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".
0
 
LVL 1

Author Comment

by:NemCert
ID: 26165134
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>
0
 
LVL 1

Author Comment

by:NemCert
ID: 26165171
Am not enable the STUN
0
 
LVL 16

Expert Comment

by:memo_tnt
ID: 26165345
that's good,, but it needs to register
so,, you need to check register with domain and receive incoming calls
in your xlite

0
 
LVL 19

Accepted Solution

by:
feptias earned 500 total points
ID: 26165765
"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.
0
 
LVL 1

Author Comment

by:NemCert
ID: 26172561
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
0
 
LVL 1

Author Closing Comment

by:NemCert
ID: 31670600
Thanks Mr.Voip ..
0

Featured Post

Eye-catchers on the conference table

Challenge: The i-unit group was not satisfied with the audio quality during remote meetings. They were looking for a portable solution with excellent audio quality for use in their conference room but also at their client’s offices.

Question has a verified solution.

If you are experiencing a similar issue, please ask a related question

The Zaptel people (www.zaptel.com) got kind of annoyed with the fact that they were getting bombarded with searches for the zaptel driver system for Asterisk (not to mention they own the trademark on zaptel). So, they kindly requested that Digium ch…
Skype is a P2P (Peer to Peer) instant messaging and VOIP (Voice over IP) service – as well as a whole lot more.
Please read the paragraph below before following the instructions in the video — there are important caveats in the paragraph that I did not mention in the video. If your PaperPort 12 or PaperPort 14 is failing to start, or crashing, or hanging, …
Is your OST file inaccessible, Need to transfer OST file from one computer to another? Want to convert OST file to PST? If the answer to any of the above question is yes, then look no further. With the help of Stellar OST to PST Converter, you can e…
Suggested Courses

829 members asked questions and received personalized solutions in the past 7 days.

Join the community of 500,000 technology professionals and ask your questions.

Join & Ask a Question