X-Lite - SIP softphone

3.1.3. X-Lite - SIP softphone

To register an user on this SIPclient-supporting phone we will use the X-Lite phone. You can download it from http://www.xten.com/. You have to be sure that you have already installed Asterisk and registered the users and extensions that we did.

When you download the phone it looks like this on the picture below: Do not worry about the error message that appears on the top of the screen. Just click on the Menu button - left from the Clear button below the screen. Then go to System Settings=>SIP Proxy=>Default. There you have to adjust the phone to the user which is registered in sip.conf on asterisk.

Enabled must be switched to YES - otherwise nothing will work even if your user registration is correct. Display Name may be whatever you want to be displayed. Username and Authorization User must be as in sip.conf and in our example it is ivan. Password has to be same as secret which is set in sip.conf for the user.
Domains and Proxy have to be the IP addresses of your asterisk server. Note that here my asterisk IP is 10.3.3.25 but in your case it might be different.

Now when user ivan or anybody else from the context dials 4321 the user test will be called.

Register the user on X-Lite in the same way like the user ivan.Go to /Start Menu/System Settings/SIP Proxy/Default. The password is again same as the secret in sip.conf where user test is registered.

X-Lite cannot use silence detection when it is used with Asterisk server. To solve this problem we must turn on the 'Transmit Silence' option. We can find this option in Menu>Advanced System settings>Audio Settings>Silence Settings>Transmit Silence. By doing this we are assured that X-lite will send always audio frames even after it detects a silence.

Now we have two registered users who can call each to each. The Command Line Interface (CLI) can be access by typing in Linux: safe_asterisk and then the asterisk "r"
Remember to type reload in CLI every time you change a file in Asterisk.

For information how to make the configurations in the Asterisk's configuration files please read our tutorial about the Configuring IP Phones for use with Asterisk

So, when you open the X-Lite phone where user "ivan" is registered and call number 4321 the other user - "test" on the other pc will be called. Here is what will you see in CLI when doing this.

User Comments

bwwooo223 (svenska dot hospitalet at maria dot inkA)
30 December 2008 23:58:24
hyyu bytt ouy tellephnone fyrt i-phone?'
helsnih gyyn tr'y ir cont batt olny minuten gui
knatty mottyy bor tellephnone fyrt anm anther lijk sony ericcson model:4355 snu? muy shild canat wajt onne tayt isnok mty knopp dre py i h HELSNGH:MARIA tyuy

Kirti (Kirti dot prajapati_it at yahoo dot com)
27 December 2008 05:10:37
my x-lite software can not work proper .when incoming call came but outgoing not response.

vijay kumar (vijayb4u dot kumar at gmail dot com)
25 December 2008 14:33:15
Hi all,
I installed the x-lite softphone in my windows XP pc and configured as follows:
Display name: xxx
User name: xxx
Password: xxx
Authorization name: xxx
domain: nnn.nnn.n.nn
proxy address: nnn.nnn.n.nn:5060

where nnn.nnn.n.nn is the address of my linux box where I installed Asterisk.

I also installed the x-lite softphone in the same linux box where I installed Asterisk and configured it the same but instead of xxx, i used yyy

This is the sip.conf in the linux box:
[xxx]
type=friend
secret=xxx
host=dynamic
canreinvite=no
context=internal

[yyy]
type=friend
secret=yyy
host=dynamic
canreinvite=no
context=internal

The extensions.conf has the following section:
[internal]
exten => xxx,1,Dial(SIP/xxx,,r)
exten => yyy,1,Dial(SIP/yyy,,r)

This is the problem:
If I call extension yyy from the phone installed in the linux box, it dials and calls itself (i'm able to answer on the second line)

If I call extension xxx (phone installed in Windows) from the yyy extension (phone installed in the linux box), it fails (call failed: 4004 not found).
The same happens when I called from the phone installed in Windows. In other words, I cannot make these two phones talk to each other.

if I enter in CLI the command "sip show peers", it shows both extensions, xxx and yyy but the only one on-line is yyy (the one in the same linux box). somehow it does not see the one installed and running in windows.

Both Windows and Linux are running in my home network (bellsouth homenetworking DSL) and i'm able to ping both boxes so they see each other in the network.

Any help will be greatly appreciated. I'm very new to this an I'm struggling to understand how it works.

Vincent (summarch22 at yahoo dot com)
19 December 2008 15:08:13
I'm having a problem with my the audio setting. I can hear them but they can't hear me. Please advise me as to the settings for audio. Thanks

yasso (dream7056 at hotmail dot com)
12 December 2008 15:49:58
tgjh hgjh

Claudia G (claudiagorritti at hotmail dot com)
10 November 2008 19:21:38
Hi, I'm having this problem for a few days now... If I made a free call, there are no problems... called phone rings, I can hear them and they can hear me...

If someone calls me via the break in numbers there are no problems... my phone rings, I can hear them and they can hear me...

If I call any international number... the other side rings and can hear me... but I can't hear anything... no rings, no notification of my balance or anything...

Please help me.
Thanks

jagdish (dhaliwal dot jagdish at yahoo dot com)
09 November 2008 23:04:23
it says registration error so what should i do

puneet singhi (vikash_jammy at yahoo dot co dot in)
26 October 2008 10:36:28
i have software of X-lite but how can i oprate this software. please give me any suggestion

ljh (levihoward dot 18 at gmail dot com)
26 October 2008 08:26:24
hi
i am from australia
over here we have special '1800' and '1300' nubers for easy calling
anyone know how to call these with x-lite
cheers

ankit (ankit_varshney at yahoo dot com)
21 October 2008 16:57:44
after downloading xlite,how do i get the username and password details and how do i activate my username to make calls using xlite?

amier (amiersaid at yahoo dot fr)
13 October 2008 15:39:27
pas de commentaire

Edilson (djedilson at hotmail dot com)
09 October 2008 18:43:41
gostaria do tutorial X-Lite SIP VoIP Softphone

GIRISH (giri dot manukula at gmail dot com)
07 October 2008 11:32:18
can anybody tel me how to connect from trixbox or asterisk software to X-lite softphone and BOL SIP phone. Please send me suggestion

GIRISH (giri dot manukula at gmail dot com)
07 October 2008 11:27:53
can anybody tel me how to connect from trixbox or asterisk software to X-lite softphone and BOL SIP phone. Please send me suggestion

GIRISH (giri dot manukula at gmail dot com)
07 October 2008 11:24:28
can anybody tel me how to connect from trixbox or asterisk software to X-lite softphone and BOL SIP phone. Please send me suggestion

nayeen (nayeen_al_amin at yahoo dot com)
01 October 2008 10:24:17
hello

marie (mah2_06 at hotmail dot fr)
15 September 2008 10:32:29
slt si kelkun peu maidé
je veux faire une connexion entre deux x-lite
un sous windows et l'autre sous linux
avec un serveur asterisk sous linux ausssi
est c kil y a kelkun ki peu maidé
merci

Sunny Sunshine (shahzad_sunshine at yahoo dot com)
05 September 2008 21:37:51
Hi , I'm Sunny from Dubai Like to make an Account , But I cant get any Opposition in any site , Could you help me ? Plz ? Thanks Regards .

Evans Tallam (ektalama at yahoo dot com)
04 September 2008 08:12:21
i cant hear no sound when i dial
i don't the problem

akash (krjason145 at yahoo dot com)
03 September 2008 22:02:11
i do have the software bt dont have a sip account and domain and password so plz help me

kashif (kashif_78300 at hotmail dot com)
26 August 2008 02:22:19
i have user and pasward for wateen plz give me proxy ip halp

jp (jprakashuae at yahoo dot com)
22 August 2008 16:10:16
i am download the xlite softphone but i cant use it becouse i have no username, password and other things so pls help me.

Sougata Pal (sougata dot pal at gmail dot com)
19 August 2008 14:20:33
I have download X-LITE but strangely , when i am going to find menu button i cant find in my downloaded version. When i check some sites everywhere i see its near the clear button but strangely my downloaded version dont have any clear in that place. The clear button is below and there is no menu button.

Nediank (nedian2k at hotmail dot com)
18 August 2008 15:36:37
Hello i can receive a call from outside but i cannot do outside call and i have checked completely in outbound ,i am in UAE could you please guide me in Tribox asterisk..

fazal (fazals at MSN dot COM (fazals at msn dot com)
17 August 2008 10:44:30
i just install X lit and i dont know how to configure it.

i try to call my home country Pakistan from Doha Qatar using X-lit. will appriacialt if any one help me in this regardss

Fazal

Guillermo Méndez Avila (gmendez123 at hotmail dot com)
07 August 2008 04:44:55
Deseo probar la última version.

hussien m (hussien3 at gmail dot com)
05 August 2008 09:22:31
hi , frined 00962 79 5068605

irfan (fanu86 at yahoo dot ca)
21 July 2008 22:08:02
any one can tell me if I can use phone set(usb phone etc) x lite is perfect cool with voipgo.com

mario (bootstomp_mario_77 at yahoo dot com)
11 July 2008 19:17:22
good

Spider (alaa_eldin20052003 at Yaho dot com)
08 July 2008 21:27:36
aaa

jabbar (jabbartelecom at yahoo dot com)
07 July 2008 19:15:20
friend

Ali Shoukat WateeN Telecom (ali dot Shoukat at wateen dot com)
07 July 2008 15:28:55
we need this on good way where we can resolve lot of problem if it work any body find any problem and then its resolution he should contect us at 0092-0321449674 thank you
ali shoukat

humayoun khan (humayoungandapur at yahoo dot com)
16 June 2008 19:47:17
i have download softwear and i pay the mony for credit 10 $ .but i dont know my mony where gone.so plz whats the procedure of this making call and where is my credit ..i am waiting for response

Dexter (dexdexter at hotmail dot it)
30 May 2008 17:41:31
I have x-lite. It works very well...

jean (zahs_12 at yahoo dot fr)
21 May 2008 16:28:02
je suis proprietaire de ce telephone

hooo1 (dongnampl at yahoo dot com)
21 May 2008 07:46:57
how to sing up

Mir (kjoush at yahoo dot com)
18 May 2008 17:29:17
Please Help!

I downloaded X-Lite Voip phone from internet. I dont know how to use it. Also I dont know how to make SIP account and what to put in the field to make my SIP account. I heared that X-Lite give one number to us and we can give this number to back home so they can call us free? Is that true, if yes please help me to configure all setup step-by-step.

I will appreciate.

Thank yo very much.

regards,

Mir

rashid mustafa (rahidmo at yahoo dot com)
17 May 2008 15:48:37
rashid

mohammd (never_4never at yahoo dot com)
09 May 2008 06:36:48
very good

mulkamon (macsim-80 dot 80 at bk dot ru)
03 May 2008 07:55:07
I want to link the softphone

N1ka (xz at xz dot com)
29 April 2008 18:39:09
lol !.. Login failed! Contact Network Admin! < lol p.s you must have a host where you can register.. if dont want then download tribox.. and in proxy write your IP adress.. newbie lol..

Chitra (chitra26jan2002 at yahoo dot co dot in)
13 April 2008 21:06:18
I want to downdoad

Chitra (chitra26jan2002 at yahoo dot co dot in)
13 April 2008 21:05:35
I want to downdoad

Vamsidhar (vamsidhar dot reddys at gmail dot com)
13 April 2008 04:35:56
whats this registration error-404 not found???

rupa (swar_rup at rediffmail dot com)
19 March 2008 20:49:25
ITz jus kool !

sandeepkg (sandeepkumar dot tech at yahoo dot com)
19 March 2008 12:57:27
when i want to make a call it say to me inter the pin number
how i take it and how i feed

nerman (mani_2006nerman at yahoo dot com)
12 March 2008 18:42:23
thanxxxxxxxxxx

bernie martin (Berniemartin24 at gmail dot com)
03 March 2008 18:24:34
the phone will ring but i cant hear the callerI also cant make calls on it

avtar singh (mrtattla at rediffmail dot com)
02 March 2008 09:21:25
dear sir & madam i have use tpad.com software in my liptop window vista but erro.so i today use xlite software download my computer.
user name-tpad
pass-secure
autho-sip.tpad.com but geeting erro not outgoing call solve this problem sir thx u.

Brian Berring (briancompaney13 at yahoo dot com)
15 February 2008 16:49:45
I will like to get internetphone line

aqeel (aqeeln at yahoo dot com)
11 February 2008 13:27:00
nathing special

vivek (vivek dot rathore at indscpae dot com)
11 February 2008 11:34:55
no commment

selçuk erol (selcukerol74 at hotmail dot com)
10 February 2008 14:22:05
WEFWEF

aqeel (aqeeln at yahoo dot com)
08 February 2008 10:26:07
hi
need x_lite configuration

aqeel (aqeeln at yahoo dot com)
08 February 2008 10:26:04
hi
need x_lite configuration

salah (saladin19 at bect dot com)
25 January 2008 06:11:15
lets see

umakanth (umakanth_ece at yahoo dot co dot in)
23 January 2008 14:39:39
hi

nadeem syed (nadeemnsn at yahoo dot com)
20 January 2008 02:03:15
i cant make and recieve call from x lite .
need help to confugrate x-lite

nadeem syed (nadeemnsn at yahoo dot com)
20 January 2008 02:02:00
i cant make and recieve call from x lite .
need help to confugrate x-lite

meenal (meena123 at yahoo dot co dot in)
17 January 2008 08:54:25
Good site man. Helped a lot

prashant (prashantpanwar at gmail dot com)
11 January 2008 13:49:36
I got this error "registration error 407 proxy authrization" please help me out

Rajeshwar Dayal (rdayal2002 at yahoo dot co dot in)
02 January 2008 15:37:10
I've got Asterisk. When I try to dial either phone I get "call failed: Request Timeout". Any ideas ? Can anybody tell me how to get it . I don't know where I m going wrong.
Thanks !

Regards
Dr. Dayal

Jason Ginter (Jason at planettelecom dot com)
29 December 2007 23:42:48
Is call transfer funcitonality available on the Xlite IP softphone between 2 users in the same office?

Does it require an upgrade, or product registration; & is there a fee for a version that allows Xlite users to transfer to another Xlite softphone extension?

Buddhi Kunwar (buddhikunwar at yahoo dot com)
16 December 2007 08:19:18
Dear Sir,

I am regular user of Tpad. I purchased USD20 credit on 15 December 2007 and it has already the amount from my account. But when I tried to connect the call from Tpad softphone, it appears as "time out". Please suggest me to solve this problem. I could not make a single call after I purchase credit. This is an urgent request from my side. Regards.

Brian (brian at circlefusion dot com)
14 December 2007 02:20:52
The newest X-lite doesn't look like the pictures shown. Could you update the tutorial to reflect it? I'm not sure where to find all of the options shown here.

الموردالقريب (ahmed_bokhary at yahoo dot com)
29 November 2007 16:35:22
aqvcc

GM (gmontano76 at yahoo dot com)
28 November 2007 00:45:47
sir good day i got a Registration error 408 how could i correct this.

thanks in advance

Mazher Yazdan (mazdan_pk at yahoo dot com)
26 November 2007 19:56:55
test

ipunk (ipunk_2002 at yahoo dot com)
15 November 2007 10:19:11
sir how to confing x-lite just for PC1 to PC2.thanks

doron (doront1 at zahav dot net dot il)
02 November 2007 01:51:10
jhw

STEPHEN (STEPHEN dot SWIFT at ATLAS-COMMS dot COM)
31 October 2007 16:58:20
I HAVE PROBLEM WITH MY XLITE, I CAN HEAR THE OTHER END BUT THEY CANT HEAR ME. THE OTHER END BEING A VOIP SET OF A MITEL PHONE SYSTEM. CALLS TO ANALOGUE SETS ARE OK THOUGH. AND IDEAS

MANY THANKS

sahil (sexysahil21 at rediff dot com)
15 October 2007 22:11:32
hi , sir i got 401 error ...as unauthorisized user ....whats the prob is this ....

Rodrigo (brzapa at gmail dot com)
15 October 2007 20:38:24
Congratulations, helped me a lot!

BarryM (barry dot mccall at gmail dot com)
08 October 2007 07:40:35
I've got a cisco 7940 upgraded to SIP 8.8 that works fine. When I try to get a softphone client to work with my Asterisk PBX I get the following error,
Oct 8 06:40:01 NOTICE[11366]: chan_sip.c:11296 handle_request_register: Registration from '<sip:xlite1@192.168.200.10>' failed for '192.168.200.49' - ACL error (permit/deny)

This happens with all the softphone clients I try. XLite, ExpressTalk, etc... I've looked everywhere hoping to find info and cant find anything. Any help would be great.

fssg (drcowood at msn dot com)
30 September 2007 16:11:09
dsfaf

ARD Prasad (ardprasad at hotmail dot com)
28 September 2007 16:57:58
Hi,
I used x-line in my office and it is working fine. Later, when I loaded on my home system, I was not able to call other number though I could get incoming calls. How do I correct it.
Thanks in Advance
ard

hardik (cool_hadrik at yahoo dot com)
22 September 2007 15:29:20
hi , its not working. plz send me domain n all things which is used to work X-lite. I sure that i will use for my good work.

Justin (jnsoagam at yahoo dot fr)
15 September 2007 15:22:35
xvdf

siva (sivaprasad_be2000 at yahoo dot com)
10 September 2007 08:30:57
hi, i am siva. i have instaled asterisk in one linux pc, i got extensions.conf and sip.conf;
then i installed xlite in 2 windows pcs.
i have configured the extensions.conf and sip.conf with the both xlite ids.
i have added the contacts in xlite also, but i am unable to call to other xlite user.

suraj (suraj at staff dot ownmail dot com)
30 August 2007 16:47:51
Hi,
This is suraj and I we are using asterisk ipbax.For security purpose I want to disable caaling out from sip and wants agents to just make calls.Please can anyone help me out in that.

vinod mall (mallvinod at hotmail dot com)
29 August 2007 11:24:21
when i start the phone I get the message "network not connected ". After that I can not get through. Will someone pl help me out.

Vinod

morara (morarabau at yahoo dot com)
24 August 2007 13:39:52
please i cannot log in to my softphone(x-lite).my username is 1398721.please help

janasis (amitjanasis at gmail dot com)
19 August 2007 20:53:13
we want only testing?

Sreeraj TR (sreeraj1237 at gmail dot com)
16 August 2007 17:57:13
hai...........

Sheyla Jones (sheyla9623 at hotmail dot com)
03 August 2007 19:43:05
We use phones Xlite 3.0 and we are having issues with the incoming calls.... we can not hear the other end... they can hear us just fine.
Can u please tell me what to do

Best Regards

Sheyla Jones

r amesh (ramesh dot mimit16 at gmail dot com)
01 August 2007 19:32:21
this is an good step towards the digital telephoy.

AMINE (tomcrous39 at yahoo dot fr)
28 July 2007 13:17:03
thanks

paulo giovani (paulogiovani_pg at hotmail dot com)
26 July 2007 16:38:55
ñ sei usar com faço?

aa (aa at 11 dot com)
26 July 2007 10:48:13
1

yazdansk (yazdansk at gmail dot com)
17 July 2007 19:12:01
i need your help and could please make for me SIP account

xasan (olol_170 at yahoo dot com)
16 July 2007 07:21:29

From: Lee <sip:1000@192.168.81.129>;tag=2165156588
To: Lee <sip:1000@192.168.81.129>;tag=as570d31ea
Call-ID: FE4F064D133F42DA8E9C34C1687EBA84@192.168.81.129
CSeq: 51397 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:1000@192.168.81.129>
Content-Length: 0
a1fa (a1fa at nomail dot com)
25 December 2005 10:53:05
Meghna,

belione (belione12 at yahoo dot fr)
08 July 2007 19:45:00
salut

JORGE S (mexfrgroup at gmail dot com)
07 July 2007 03:31:05
VERY HAPPY WITH THIS SOFTPHONE AND WITH www.voiptogoextreme.com , with whom I am making long distance calls very very inexpensively from Mexico, in Mexico to all the world. Jorge S.

FUAAD (MEAN_2003 at HOTMAIL dot COM)
25 June 2007 20:10:55
HI ALL THE MANEGERS OF THE PHOTE I WANT TO HELP ME I WANT TO CONNECT TO MY FAMILY WHO LIVE IN AFRICA I WANNA FREE CALL THANK YOU

thameur (ismailthameur at yahoo dot fr)
16 June 2007 22:50:32
hi;
i using for asterisk-1.4.0 with fedora core 6 to configure too subscriber X-Lite, and it show for me an warning as "can not get own IP address, SIP disabled" and when i try to telnet one subscriber on port 5060 it couldn't work.
please help me to do my project as soon as possible.
thanks for all.

tadjidine (tadjidine_20 at hotmail dot com)
10 June 2007 23:33:57
I'm a student in cairo-egypt and i am from comoros.
i'd like to know how can I make call for free in this site
and i hope you will help me about that

goly (kitkdngo at yahoo dot com)
30 May 2007 14:11:07
download

leonard (nadi-peja at hotmail dot com)
29 May 2007 00:37:39
hello x-lite

giorgi torelishvili (gogitpress at yahoo dot com)
19 May 2007 00:34:25
please help me to register X-lite in sip accounts

umarkhan (moghul_shaik at yahoo dot com)
13 May 2007 09:22:22
Hi to x-lite org.This feature will capture each and every consumer on planet.

dori (dori dot starosa at gmail dot com)
02 May 2007 07:33:18
I need to see whose clients number are calling in. How do we do about this? is there an alternative?

PM (mehtap5 at netscape dot net)
26 April 2007 12:50:49
I have three location with Trixbox installed and trunks added between each. My problem is that i am unable to see the online status of Xlite users from other locations. I can only view online status for my local users. Is there a way with which i can view online status/availability of users from other locations? Please help..

PM

mostafa (one_secret1 at yahoo dot com)
23 April 2007 23:36:34
thankx

hameed (nau_shi at yahoo dot com)
21 April 2007 17:10:29
dearest supplier i need this card.plz provide this card as early as possible.

Shrek (haijiangxiaoxin at yahoo dot com dot cn)
19 April 2007 10:10:12
thank you for your example, it's considerable useful

naser (naser_140_66 at hotmail dot com)
16 April 2007 20:45:39
naser

cabdixakiin (cabdixakiin30 at hotmail dot com)
12 April 2007 13:55:21
I am happy boy

roshan (roshan75lk at gmail dot com)
09 April 2007 09:41:48
sS

dmnfg.edf (belione12 at yahoo dot fr)
30 March 2007 22:31:19
klejrtglwertf;egtge g

khalifa (mahoza2000 at yahoo dot fr)
29 March 2007 23:57:05
hi

Ravi Sajjan (ipbx dot support at gmail dot com)
26 March 2007 12:56:18
Hi Everyone,
I have configured one Asterisk based Trixbox. Now i want to make conference call acording to call center process.

With the xlite phone i am able to make conference call but if i have one FXS Audio code than ho can i make thirdparty conference for international call.

Can anyone help me out for this.

Thanks

If you have any comments or solution for that. please send me your replay in my mail address: ipbx.support@gmail.com

Thanks & Regards,

Ravi Sajjan
ipbx.support@gmail.com

moi (nafoundza at hotmail dot com)
26 March 2007 10:01:56
very good

christopher (christo_2010 at yahoo dot com)
24 March 2007 00:45:25
lil chris4ever

Jeff (alvar_4786 at yahoo dot com)
13 March 2007 06:26:36
test mail

Jeff (alvar_4786 at yahoo dot com)
13 March 2007 06:22:00
test mail

Ryan Stille (ryan at cfwebtools dot com)
09 March 2007 18:30:51
This guide must be based on an older version of X-lite, because my menus don't look anything like the screenshots. All I can get is '503 - Service Unavailable'.

souvik (souvik_sadhu at yahoo dot com)
07 March 2007 05:08:41
How do i make a call from a x-lite to asterisk land phone,which is connected to FXS?

thanks in reply.

souvik

reloded (reloded at gmail dot com)
02 March 2007 07:06:38
Hi guys. Am new to asterisk. I've followed the guides here step by step and all's good so far.
However, trying to dial from x-lite, I get:
"Call failed: Not Found"
No error code is given. However, the x-lite logs its showing:

SIP/2.0 603 Declined

On the CLI on asterisk, I get:

[Mar 2 08:52:20] WARNING[11507]: app_dial.c:1062 dial_exec_full: Dial argument takes format (technology/[device:]number1)

Someone please advice. I've been searching for a solution for the past 3 days with no success.
Hi guys. Am new to asterisk. I've followed the guides here step by step and all's good so far.
However, trying to dial from x-lite, I get "Call failed: Not Found" No error code.
On the CLI on asterisk, I get
[Mar 2 08:52:20] WARNING[11507]: app_dial.c:1062 dial_exec_full: Dial argument takes format (technology/[device:]number1)

Someone please advice. I've been searching for a solution for the past 3 days with no success.

Mahamoud (maka_issa at hotmail dot com)
02 March 2007 00:51:53
hi am ok

bobwilliams (bhemanth83 at gmail dot com)
26 February 2007 23:24:00
jnfgkl

sniderre (snider266 at hotmail dot com)
26 February 2007 19:54:36
HiIam aliaban

mahamoud (henry14 at hotmail dot com)
26 February 2007 12:27:04
jjjjjjffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbcccccccccccccccccccccccccbbbbbbbbbbbbbbbbbbbbcccccccccccccc xxxxxv aaajjjjjjjjjjjjjjjjjjjjjjjjjjjjj

mahamoud (henry14 at hotmail dot com)
26 February 2007 12:26:54
jjjjjjffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbcccccccccccccccccccccccccbbbbbbbbbbbbbbbbbbbbcccccccccccccc xxxxxv aaajjjjjjjjjjjjjjjjjjjjjjjjjjjjj

mahamoud (henry14 at hotmail dot com)
26 February 2007 12:26:15
jjjjjjffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbbcccccccccccccccccccccccccbbbbbbbbbbbbbbbbbbbbcccccccccccccc xxxxxv aaajjjjjjjjjjjjjjjjjjjjjjjjjjjjj

mahmoud (mahmoud_salah62 at yahoo dot com)
25 February 2007 15:34:38
iwant to download x-lite

aiki (laden_lee at yahoo dot com)
23 February 2007 12:57:01
good call

pitik (pit_ik at yahoo dot com)
23 February 2007 08:07:26
may be we can get freephone with x lite
tank for you if like that

oke- - - - -

omer (umer_javaid_ at hotmail dot com)
17 February 2007 13:14:54
Hi all,
I installed the x-lite softphone in my windows XP pc and configured as follows:
Display name: xxx
User name: xxx
Password: xxx
Authorization name: xxx
domain: nnn.nnn.n.nn
proxy address: nnn.nnn.n.nn:5060

where nnn.nnn.n.nn is the address of my linux box where I installed Asterisk.

I also installed the x-lite softphone in the same linux box where I installed Asterisk and configured it the same but instead of xxx, i used yyy

This is the sip.conf in the linux box:
[xxx]
type=friend
secret=xxx
host=dynamic
canreinvite=no
context=internal

[yyy]
type=friend
secret=yyy
host=dynamic
canreinvite=no
context=internal

The extensions.conf has the following section:
[internal]
exten => xxx,1,Dial(SIP/xxx,,r)
exten => yyy,1,Dial(SIP/yyy,,r)

This is the problem:
If I call extension yyy from the phone installed in the linux box, it dials and calls itself (i'm able to answer on the second line)

If I call extension xxx (phone installed in Windows) from the yyy extension (phone installed in the linux box), it fails (call failed: 4004 not found).
The same happens when I called from the phone installed in Windows. In other words, I cannot make these two phones talk to each other.

if I enter in CLI the command "sip show peers", it shows both extensions, xxx and yyy but the only one on-line is yyy (the one in the same linux box). somehow it does not see the one installed and running in windows.

Both Windows and Linux are running in my home network (bellsouth homenetworking DSL) and i'm able to ping both boxes so they see each other in the network.

Any help will be greatly appreciated. I'm very new to this an I'm struggling to understand how it works.

adeel (adeelumer86 at gmail dot com)
17 February 2007 12:45:29
i have configured xlite with asterisk but not able to make calls please email me configured sip.conf and extension.conf file to my email

yassir (yassirgo at yahoo dot com)
17 February 2007 11:25:30
ghuhjoijki

fateen baig (paradyn-1 at hotmail dot com)
16 February 2007 21:25:21
send me configuration of xlite with g729 in asterisk.

said (said_83 at inbox dot ru)
09 February 2007 07:01:03
salom

raza (raza at yahoo dot com)
07 February 2007 19:36:00
fgtf

bharathi (srsbharathi at gmail dot com)
27 January 2007 09:39:12
please tel ow to regester x lite

assad (ahumza at gmail dot com)
26 January 2007 21:59:08
hh

jam (pele_20062000 at yahoo dot com)
25 January 2007 11:17:17
my x-lite phone was not registered. what i do.

Zeeshan Sajid (shani_707 at yahoo dot com)
23 January 2007 12:04:47
It is too good ... :)

Pierre (email at domaine dot com)
18 January 2007 13:38:45
you can also try GoSIP (http://www.go-sip.com)
it's a new wifi sip softphone...

tarik (moi at moi dot tot)
09 January 2007 18:21:28
http://www.xten.com/index.php?menu=download

tarik (moi at moi dot tot)
09 January 2007 18:19:12
http://www.xten.com/index.php?menu=download

tareq (tareqdipon2003 at yahoo dot com)
07 January 2007 21:04:06
please give me a download link

fejsal (johnlushi at yahoo dot com)
23 December 2006 21:57:35
hi you there i like jast like to say that you are gret so in the time futre i hope the will be beter /. sincerly yours

Akmal basha (send22basha at yahoo dot com)
18 December 2006 13:08:34
sM

Merinch (m_yuseinov at econt dot com)
11 December 2006 08:42:07
Hi how i can disable Line 2

shivbhanu (shivbhanu18 at yahoo dot com)
09 December 2006 06:49:06
i m using x-lite software ,but getting problem like ,when i get call able to hear clearly but my voice is not going there clearly please help me

essam (essam500000 at yahoo dot com)
30 November 2006 17:10:45
i wanted to dawenload x-lite

Murtaza (murtaza_m_ali at hotmail dot com)
22 November 2006 19:29:16
like it very much

rus_baban (rus_baban at mail dot ru)
13 November 2006 08:02:28
hi.
i have problem when transferring incoming call to another user.only one phone in our net do not support this option!!!why it is so???

Sasha (arc_sy at yahoo dot com)
06 November 2006 22:10:53
I am using X-Lite phone.After i call into a 1800 number ( conference call ) I need to enter a passcode to get into the call. However the x-lite keypad does not recognise any keystrokes once the phone number has been keyed in.Is there some setting that i can change to enable the softphone to recognise the numbers that i key ?? please help. Respond to the email id listed

bangoura (alseny26 at hotmail dot com)
06 November 2006 19:03:28
je la relation avec vous .

arshad (samir_in21 at yahoo dot com)
06 November 2006 11:51:29
hi

DENIS (denislotsu at yahoo dot fr)
27 October 2006 12:28:13
Hello,
I have downloaded your soft but I'd like to know how to configure it and to use it. I have a broadband connection to have access to internet. What do I need more? Please kindly send me the answers to my email box. Thank you for your cooperation.

hibonuura (smuniir at yahoo dot com)
26 October 2006 22:28:50
In order to prevent automatic posting on our website, we kindly request you to type in the number you see in the picture below.
Image Verification:

lola (lola10010 at yahoo dot com)
23 October 2006 04:22:56
hiiiiiiiiiii

harsh (harshdeepsabharwal at gmail dot com)
13 October 2006 16:02:20
GGG

usher (robin_mox at yahoo dot com)
10 October 2006 21:03:43
that right

Souef (abdouchakour25 at hotmail dot com)
04 October 2006 23:21:57
I am very intersted with your cervice

john smith (john_smith2005 at uymail dot com)
03 October 2006 15:58:20
i wish to use your software product by pc to phone.

scorrorus (scorrorus2000 at yahoo dot com)
03 October 2006 08:02:18
GOOG DAY OVER THRER WISH TO BUYING YOUR SOFTWARE PHONE PRODUCT OK KINDLY SALE FOR ME OK STAY BLESS

saleh (abbasalehben at yahoo dot fr)
02 October 2006 20:22:59
Is it possible to transfer a call with X-lite and asterisk?

ali (dosti184 at yahoo dot com)
30 September 2006 19:43:14
hi to all.
my qustion is how to activate the hold and conference button on the
xlite soft phone....... ...can dtmf mode support to captur the
text written on keypad if yes then how to map that text to the
extention... ........i m waiting for rply,bye to all

joe (ab4nk_jo3 at yahoo dot com)
26 September 2006 00:00:56
i have problem with x-lite...please tell me how to use x-lite

ahmed (power_1200 at hotmail dot com)
25 September 2006 02:57:38
llllllllllllllllllllllllllllllllllll

ahmed omar (axmeddinho at hotmail dot com)
24 September 2006 23:53:39
hi i want send to x-lite

ghulam sakhi (sakhipwj at yahoo dot com)
22 September 2006 08:56:37
dear sir or madam!
hope you are fine and all doing well, i am from afghanistan and i hope i have a close commonecate with you,
at the moment i have a PCO and i want use your services if possible.
thank you very much
sakhi from Mazar

saritha (saritha dot naraharisetty at gmail dot com)
21 September 2006 11:35:50
Hi i configured an asterisk server and our proprietary box with voip running over it.From two different boxes i am able to make calls to each other using asterisk server as the proxy.But when i dial some number say dial 200 from asterisk server the other phone is ringing and rtp streaming is also coming.I could see the rtp packets in etereal but voice is not being heard.Can somebody help me out on this.Awaing a sooner reply
Thanks in Advance
Saritha.

abukar (abukar003 at hotmail dot com)
20 September 2006 16:12:04
hello i am abukar

kaka (prince_nhl_rien at hotmail dot fr)
16 September 2006 15:01:48
moi si je veux je peux

plus (farah_atman at hotmail dot fr)
16 September 2006 14:53:05
pour la communication avec le monde

andi (andikasp2000 at yahoo dot com)
14 September 2006 20:14:28
sok..

umesh (umeshmohanty75 at yahoo dot com)
12 September 2006 09:30:09
good

sabik (sabikva at hotmail dot com)
08 September 2006 02:57:49
ss

Mike (sorry at noway dot com)
15 August 2006 04:21:44
works like a champ!!

ShivaSankar (shiva at neopackets dot com)
10 August 2006 06:44:36
I am using Asterisk 1.2.10 and X-lite.I have implemented caller Id,Music On hold and etc....in my system but I fail to implement the call transfer.I have set the diaplan Ok(I think) but I am not able to get the desirerd O/P. My Dial()in extension.conf is executing perfectly but I have defined Transfer followed by that but that is not executing , just call is able to established.
Can any body will help me in this regard, I am a new to Asterisk.
Yah, Point to be noted that the button Transfer and Conf in the buttom of X-lite is Deactive(I dont know why).
I am also available at "shiva_pd1983", its my yahoo IM.
Expecting a sooner reply,
Thanks in Advance,
Shiva

SalfranCL (salfrancl at yahoo dot es)
30 July 2006 12:32:05
I want to share my experience.

I'm configuring in Asterisk:
1- Two SIP and IAX2 extentions with voicemail for two persons (done)
2- An extension for check the voicemail (done)
3- An automatic operator to receive automagically new calls
4- An extension to measure local echo. (done)
5- Configure X-Lite to work with Asterisk (done)

I'll give to every who ask me at salfrancl at yahoo dot es the configs files.

CLAUDIA (clauxmr at gmail dot com)
26 July 2006 18:33:03
hola,a todos quisiera preguntar como hago para llamar a xtensoftphone desde consola es decir sin necesidad de darle cd xten-xlite y luego ./xtensoftphone....quisiera solo teclear ./xtensoftphone y que ya me aparezca.

muchas gracias¡¡¡

elen (elenpoghosyan2004 at yahoo dot com)
25 July 2006 20:15:42
t

paula (elskaleonardo at hotmail dot com)
06 July 2006 11:39:11
I can installed one PC - asterisk and Softphone? I can receive call but I cannot call - Softphone

Mike (lalbi77 at yahoo dot fr)
29 May 2006 16:21:07
Hello,

I'm using X-lite behind a NAT and the problem is the connection with sip protocol is ok but rtp transport is not working. I tested many option like nat=yes(or route) or qualify=yes in sip.conf but no one work.

Can anybody tell me how I can configure for using X-lite behind a NAT ??

regards

Manish Kumar Jha (bdm at envobiotech dot com)
22 May 2006 22:17:29
hi we are into outsourcing business n our main markets r america,canada,australia and uk. so please let me know about a customised package whichvl b benefecial for us thank you

ba (shaqba at hotmail dot com)
16 May 2006 10:11:21
Hello I installed(Asterisk and also Xlite .I would like to know how to configure the proxy; and also I would like to know how to start asterisk and how to change the configurations,

hgd (hgd7 at hotmail dot com)
10 May 2006 11:38:05
ihave xlite softphone and i configer it with asterisk with isp.config
iam regester 2 users 123 and 122 when icalling when from other xlite tell me 404 not fond plese tell me for more details>

Spiro (spiro79 at gmail dot com)
01 May 2006 01:07:26
Happens the same thing as Le Son Phat.
What could be the problem?
I checked the port with netstat -nap and it shows asterisk running on port 5060.
Anyone has solved this?

Le Son Phat (sonphat_le at yahoo dot com)
14 April 2006 11:15:33
I dont know Why it shows:
Login timed out! contact network Adm
Your number is :ivan
Call Timer:0:00:00
Everything was configured:(the same sources in your tutorial)

iax.conf:
[ivan_iax]
type=friend
username=ivan_iax
secret=lesonphat
host=172.28.2.5
context=tutorial
[test_iax]
type=friend
username=test_iax
secret=lesonphat
host=172.28.2.5
context=tutorial

extensions.conf:
[tutorial]
exten => 1234,1,Dial(SIP/ivan)

exten => 4321,1,Dial(SIP/test)

exten => 1111,1,Dial(SIP/ivan_iax@iavan_iax)

exten => 2222,1,Dial(SIP/test_iax@test_iax)

sip.conf:
[ivan]
type=friend
username=ivan
secret=lesonphat
host=172.28.2.5
context=tutorial
[test]
type=friend
username=test
secret=lesonphat
host=172.28.2.5
context=tutorial

ZEESHAN MANZOOR (ZEESHANMANZOOR606 at hotmail dot com)
07 April 2006 05:57:47
i wana install this plz help me out how i can do this

kupaluts yeah (siklab_666 at yahoo dot com)
02 April 2006 10:39:32
hey we have a problem too together with my nigger here sleeping, you know what, we've been doing this all night and nothings working fine you know what the problem is, you know, there is a problem something like this to all my nigger. I send this problem 100x and theres no one responsing, my problem is again is this "cannot find extensions context 'i.e. sip,john,trust and what so ever da!" common help me my niggers out there somewhere over the rainbow oh man!!! please.

good afternoon (bricedocta at yahoo dot fr)
28 March 2006 15:59:23
I would like to build a dial plan base on softphone;no hardware card.
I would like ta have :
a)one main page on witch the call arrive and then by using the instruction "Goto" i can switch to my differents "context" or department.
main page config
exten => 9,1,Answer ()
exten => 9,n,Background(submenuopts)
exten => 1,1,Goto(sale,100,1)
exten => 2,1,Goto(support,2001)

b)two contexts : sale and support.

problems :
asterisk dont know the extension of my main page
sale and support can't speek to each other
when i use only one context for example sale,i can pass call but can have my main page.

please can you help me?
it is possible to have a main page without using hardware device?

HIPPOLYTE (hfasah at yahoo dot com dot au)
25 March 2006 14:16:39
Hi all,
I am a new commer.I want to be an asterisk guru.My problem is I don't know where to start. However , I will like to start with two PCs.(Pentium 3 500Mhz 128MB, running windows XP Pro.I also want to use X-lite soft phone for a start.Please can someone give me the steps to follow,I want to impress my friends and teacher in the next 2 weeks.Should I network the PCs b4 installing Asterisk.
Waiting for your reply. Thanks

hamad (hamadalsser222 at hotmail dot com)
23 March 2006 12:10:19
how can i set my sterisk ip phone

Waiting for your reply.

Best Regards

hamad

nishanth (nissshhh at yahoo dot com)
20 March 2006 15:58:56
guys i am not able to find the source code for this softphone could someone please sen d it across to the mail address....

Simon (smonlim1 at yahoo dot com)
19 March 2006 03:17:55
Hey all, could your help. I'm a newbie.

Got 2 x-lite clients running on different computers. Both are logged in. I got 2 extensions configured on asterisk (200 and 300). I get a dial tone but when I call it (ext 200 or 300) goes no where. What am I missing? The only way I could get my soft phones to show "logged in" was to configure register=never.

ikzer (rez_inge2000 at yahoo dot fr)
18 March 2006 15:42:13
I want to know how add the second user "test" in sipphone "xlite":
1. in [Default] section of his softphone (second PC)
or
2. in [proxy1] section ?

Adrian Leonte (adrian dot leonte at evolva dot ro)
06 February 2006 12:08:41
hello, sorry for me english
here is an example, i used on me Asterisk and it work
I have 3 x-lite softphone and 1 fxs AudioCodes mp104

extension.config

[general]
static=yes
writeprotect=no
[globals]
xlite=SIP/from-sip
[default]
include => from-sip
[from-sip]
exten => 042200,1,Dial(SIP/042200)
exten => 042201,1,Dial(SIP/042201)
exten => 042100,1,Dial(SIP/042100@10.121.1.7)
exten => 042101,1,Dial(SIP/042101@10.121.1.7)
exten => 042102,1,Dial(SIP/042102@10.121.1.7)
exten => 042103,1,Dial(SIP/042103@10.121.1.7)

and the sip.config will be

[general]
context=sip
allowguest=yes
realm=10.121.1.2
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
domain=10.121.1.2
[authentication]
[042100]
type=friend
username=042100
secret=password
context=from-sip
host=dynamic
disallow=all
allow=all
;
[042101]
type=friend
username=042101
secret=password
context=from-sip
host=dynamic
disallow=all
allow=all
;
[042102]
type=friend
username=042102
secret=password
context=from-sip
host=dynamic
disallow=all
allow=all
;
[042103]
type=friend
username=042103
secret=password
context=from-sip
host=dynamic
disallow=all
allow=all
;
[042200]
type=friend
username=042200
secret=password
context=from-sip
host=dynamic
disallow=all
allow=all
;
[042201]
type=friend
username=042201
secret=password
context=from-sip
host=dynamic
disallow=all
allow=ulaw
;
[042202]
type=friend
username=042202
secret=password
context=from-sip
host=dynamic
disallow=all
allow=all

Shiza Kamal Faridi (shizakamal at gmail dot com)
29 January 2006 17:21:09
Nothing going fine.

Kamal Faridi (kamaluf at gmail dot com)
29 January 2006 17:20:39
Still facing the issue nothing resolved as yet

Kamal Faridi (kamaluf at gmail dot com)
25 January 2006 22:02:50
I am not connecting with this phone please assist.

Nishant M (nishantmdoak at gmail dot com)
17 January 2006 10:12:22
can anyone let me know the way conferencing can be done in X-lite ?

Sheeju R Alex (sheejuec7 at gmail dot com)
08 January 2006 10:27:51
Can any one tell me how to configure audio device for xlite softphone on linux. I registered a SIP user and it is working fine with the xlite softphone but I couldn't hear any voice from the caller.

Jayesh Shrivastava (jayeshshrivastava at yahoo dot co dot uk)
31 December 2005 08:53:38
Can I use X-Lite straight way over windows platform without using Asterisk within my Local Area Network and also among two long distance locations using internet.

lee (skan1603 at empal dot com)
30 December 2005 18:09:18
I can't login and receive this message.

"Login failed! Contact Network Admin"

<<sip.conf>>

[1000]
username=1000
secret=abc123
context=mytest
host=dynamic

<<extensions.conf>>
[general]
static=yes
writeprotect=yes

[globals]
XLITE=SIP/1000

[mytest]
exten => 1367,1,Dial(SIP/1000)

exten => 2890,1,Wait(2)
exten => 2890,2,Answer
exten => 2890,3,Playback(demo-echotest)
exten => 2890,4,Echo()

exten => 2468,1,Dial(${XLITE})

<<X-Lite SIP Proxy(default)>>
Enabled Yes
Display Name Lee
Username 1000
Authorisation User 1000
Password abc123
SIP Proxy 192.168.81.129
Domain/Realm 192.168.81.129
Register Always

<<X-Lite Diagnostic Log>>
SEND TIME: 9389832
SEND >> 192.168.81.129:5060
REGISTER sip:192.168.81.129 SIP/2.0
Via: SIP/2.0/UDP 192.168.81.1:5061;rport;branch=z9hG4bK3B94A845EB374991921384FE8FB486A9
From: Lee <sip:1000@192.168.81.129>;tag=2165156588
To: Lee <sip:1000@192.168.81.129>
Contact: "YourName" <sip:1000@192.168.81.1:5061>
Call-ID: FE4F064D133F42DA8E9C34C1687EBA84@192.168.81.129
CSeq: 51397 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1105x
Content-Length: 0

RECEIVE TIME: 9389929
RECEIVE << 192.168.81.129:5060
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.81.1:5061;rport;branch=z9hG4bK3B94A845EB374991921384FE8FB486A9;received=192.168.81.1
From: Lee <sip:1000@192.168.81.129>;tag=2165156588
To: Lee <sip:1000@192.168.81.129>;tag=as570d31ea
Call-ID: FE4F064D133F42DA8E9C34C1687EBA84@192.168.81.129
CSeq: 51397 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:1000@192.168.81.129>
Content-Length: 0

a1fa (a1fa at nomail dot com)
25 December 2005 10:53:05
Meghna,
The port 5060 is allready in use. Make sure you dont have a previous version running.

Run netstat -nap
You should see something like:
udp 0 0 0.0.0.0:5060 0.0.0.0:* 22508/asterisk

Meghna (meghs413 at gmail dot com)
19 December 2005 10:58:35
Hi all,

I have some problem when I launch asterisk.
I have installed the latest version of asterisk which is 1.2.1.
when I launch asterisk I get this error.
chan_sip.c:12562 reload_config: Failed to bind to 0.0.0.0:5060: Address already in use

if I configure the bindaddr=10.0.1.4 in sip.conf then also it gives
me the same error.
chan_sip.c:12562 reload_config: Failed to bind to 10.0.1.4:5060: Address already in use

Does anyone has idea what is going wrong here.

Thanks in advance.

Meghna

sandhu (sandhu159 at gmail dot com)
15 December 2005 12:12:42
how can i go on connecting local asterisk server with an extension of an Voip service provider so that i can connect my soft xlite phone to that extention of voip service provider. can ne body help me
thanks in adv

Steve (isitvacationtme at yahoo dot com)
14 December 2005 06:24:25
Not able to dial out an extension with XLite. Anybody have any ideas why?
I am able to dial into Asterisk with an outside line and even dial the extension to get to the XLite application.
But I am unable to pick dial an extension with xlite. It is as if the Asterisk server does not notice any of the keys are being pressed.

Meghna (meghs413 at gmail dot com)
09 December 2005 15:58:39
I am using g729 codec in my SIP hardphones and we have a license for g729 codec. here is my configuration for two phones in sip.conf

[321]
type=friend
username=ivan
host=dynamic
canreinvite=no
context=sip
disallow=all
allow=g729
allow=alaw
allow=ulaw

[123]
type=friend
username=ivan
host=dynamic
canreinvite=no
context=sip
disallow=all
allow=g729
allow=alaw
allow=ulaw

Now when I try to call from one phone to another the other phone rings
but conversation cannot go on.
Following are the error messages I get on asterisk CLI.

channel.c:2155 ast_channel_make_compatible: No path to translate from SIP/123-e3f8(4) to SIP/321-7774(256)

app_dial.c:1022 dial_exec: Had to drop call because I couldn't make SIP/321-7774 compatible with SIP/123-e3f8

Anyone has idea about this error? is there any other thing I need to configure or add in any other configuration files?

Meghna (meghs413 at gmail dot com)
09 December 2005 15:48:41
hello tejas

try out this. in extensions put as follows

[from-sip]
exten => 2000,1,Dial(SIP/2000,20)
exten => 2001,1,Dial(SIP/2001,20)

tejas (tejas705 at yahoo dot com)
01 December 2005 11:14:30

i have set an asterisk server and 2 sip phones(on 2 seperate PCs, all are in same network).i configured sip.conf as follows :

[2000]

type=friend
username=2000
secret=tejas
host=dynamic
context=from-sip
mailbox=100

[2001]

type=friend
username=2001
secret=tejas
host=dynamic
context=from-sip
mailbox=101

and i configured extension.conf as follows:

exten => 2000,1,Dial(SIP/2000,20)
exten => 2001,1,Dial(SIP/2001,20)

now i m getting error : cannot find extension context 'from-sip'.
can anybody help me to solve this error.

thanks in advance

tejas

Aparna (aparna_grk at yahoo dot com)
24 November 2005 15:27:56
I get the message Login Failed! contact Network Admin.
I have set up the sip.conf with

[xlite]
port=5060
host=dynamic
type=friend
secret=xlite
username=xlite
context=sip-trunk

I do not use NAT. I set up the same username and secret in X-Lite too. No console messages also appear on the asterisk console. How do i proceed ?? I dont seem to understand what is going wrong where!
Kindly help!!!

thanks n regards

ok (oklein at smallo dot ruhr dot de)
22 November 2005 06:26:13
Hi,

I connected a xlite to my asterisk and it seems to work, but i still get messages like

"*CLI> Nov 22 06:27:17 NOTICE[30132]: chan_sip.c:10815 handle_request_register: Registration from 'oklein <sip:oklein@10.128.60.157>' failed for '10.128.60.157' - Username/auth name mismatch"

but i am sure, the Username/Password is correct and i can dial other numbers. What can that be?

Hanis (noorhanis85 at yahoo dot com dot sg)
18 November 2005 12:51:35
how to make a conference call using X-lite on linux?

Meghna CHAVDA (meghs413 at gmail dot com)
18 November 2005 10:59:20
Hello,

Thanx for the support of Asterisk.

Well now I am using windows messenger 5.0 to register with asterisk.
everything works fine on asterisk side like when I do database show it
shows me the registeration of windows messenger.

But the problem is windows messenger cannot login as a SIP user, it gives me error like this: "You have been signed out of SIP Communications Server because that service has been temporarily shutdown. Please try again later."

my server address is 10.0.1.7 and I enter login as tutu@10.0.1.7 in windows messenger. do you have any idea what is going wrong? i am behind NAT. Do you have windows messenger configuration example?

Thanks.

Regards
Meghna

Humberto (humberto at saback dot com)
10 November 2005 21:12:27
How may I disable Lines 2 and 3 from X-Lite?

Lacho (support at asteriskguru dot com)
07 November 2005 18:34:37
Sorry but I did not understand the one about "the phone with no. 1010(ST2030) has dialplan |1234|" What to you mean with dialplan |1234|.

Regards

Lacho (support at asteriskguru dot com)
07 November 2005 18:31:20
Hi,

First, about the SIP - in the Dial application SIP shows the protocol(or channel), through which your phones will communicate. The configurations for this protocol have to be made in sip.conf.

Now about the context - the option context in sip.conf file, points the name of the context in extensions.conf, where you have your Dial extensions. I will show you.

Try with this configuration in sip.conf
[1010]
type=friend
host=dynamic
canreinvite=yes
disallow=all
allow=all
username=ST2030
context=dial

[1020]
type=friend
username=ST2031
host=dynamic
canreinvite=yes
disallow=all
allow=all
context=dial

And extensions.conf
[dial]
exten => 1010,1,Dial(SIP/1010,10,t) //for user ST2030
exten => 1020,1,Dial(SIP/1020,10,t) //for user ST2031

Note that both of your users have to be in one and the same context.

The meaning of the number 1010 after the arrow is, that this is the extension, which you have to dial in order to be connected the desired user. So you have to dial 1010 if you want to be connected to user ST2030 and 1020 if you want to be connected to user ST2031.

Maybe it is good idea to take a look at our tutorial about the configuration of IP phones to use Asterisk.
http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html

Meghna (meghs413 at gmail dot com)
07 November 2005 18:04:11
Hello Lacho

Thank you very much for your support.
But still it doesn't work it gives me the same error.
when I dial 1020 from the phone 1010(ST2030) then this
phone says that 1020 is wrong number and same is the case
when I dial 1010 from the phone 1020.

This is my configuration of two phone in sip.conf

[1010]
type=friend
host=dynamic
canreinvite=yes
disallow=g723.1
allow=g729

[1020]
type=friend
username=ST2031
host=dynamic
canreinvite=yes
disallow=g723.1
allow=g729

and in extensions.conf

exten => 1010,1,Dial(SIP/1010,10,t) //for user ST2030

exten => 1020,1,Dial(SIP/1020,10,t) //for user ST2031
the phone with no. 1010(ST2030) has dialplan |1234|,
does this has something to do with?

Can you tell me what does 1010 after the arrow mean and
also what is significance of writing SIP after Dial? Is
it(SIP) is the context which I have to specify in sip.conf
in the declaration of the phones.

[1020]
context=sip

like this?

I don't know where I m going wrong.

Thank you.

Best Regards
Meghna

J (cehdude at gmail dot com)
06 November 2005 23:13:29
has anyone else sniffed out the connection and noticed that the xten phone is "phoning home" to xten? I wonder if they're tunneling back into my system. It's making UDP connections back to home base. I went through the settings and tried to remove all references to xten but it still makes the connection. I guess I can restrict it at the firewall.

Are there any other SIP clients out there that I can feel safe using?

Lacho (support at asteriskguru dot com)
05 November 2005 15:56:30
Hi Meghna,

The problem is that you are trying to dial the username. You have to dial the name of the context in sip.conf. For example if you what to dial user ST2030, you have to write SIP/1010 in the dial extension, instead of SIP/ST2030.

In other words, if you change your dial extensions in extensions.conf file to:

exten => 1010,1,Dial(SIP/1010,10,t) //for user ST2030
exten => 1020,1,Dial(SIP/1020,10,t) //for user ST2031

Everything has to be alright.

Regards

Meghna (meghs413 at gmail dot com)
04 November 2005 16:47:28
Thanx for the reply. Now my hard phone registers with asterisk server.

But I am confused with extensions.conf file. I mean how should I configure this file so as to make outgoing call from one phone to another phone in the same network using asterisk?

When I call from one phone to another it gives me message Wrong Number on the phone and on ethereal traces it give Status: 404 Not Found.

Here is my configuration.

Asterisk server: 10.0.1.7
Following are my two phones in sip.conf

ip address: 10.0.1.200
[1010]
type=friend
username=ST2030
host=dynamic
disallow=all
allow=ulaw
allow=alaw

ip address: 10.0.1.205
[1020]
type=friend
username=ST2031
host=dynamic
disallow=all
allow=ulaw
allow=alaw

in extensions.conf file

exten => 1010,1,Dial(SIP/ST2030,10,t)
exten => 1020,1,Dial(SIP/ST2031,10,t)

hope you understood my problem.

Waiting for your reply.

Thanx

Best Regards
Meghna

zoa (support at asteriskguru dot com)
03 November 2005 21:50:56
host is the ip address of the phone, not the server.

Also, if you plan on having the phone register to asterisk, its best to put host=dynamic.

If it still doesnt work, post your xlite config into a post on the forum.

Meghna (meghs413 at gmail dot com)
03 November 2005 14:18:49
hello,

What will be the configuraiton of X-Lite phone
in sip.conf file?

will it appear as follows?

[xlite]
port=5060
host=10.0.1.7
type=friend
secret=thomson
username=ivan
context=Powerbook
callerid="ivan"<ivan>

here host is the ipaddress of asterisk server. is it right or it should be the ip address of x-lite phone?

whenever i run asterisk server it gives me this error
Login failed! Contact Network admin. Does anyone have idea about this?

Waiting for your reply.

Best Regards,
Meghna

Zowwie (confidential at glo dot com)
20 September 2005 22:30:38
Transfering from X-Lite w/ Asterisk.

While on a call press POUND and it should ask for an extension. If so, enter the new extension and press POUND again.

This avoids the transfer button being gray'd out on X-Lite.

From more info... Check your dial plans and other features under /etc/asterisk/*.conf

--Zowwie

ivan (support at asteriskguru dot com)
20 September 2005 10:39:19
Mike,
make sure that both pc-s use the same asterisk server. Then make sure the users you want to dial are registered. To check this in CLI type "database show". If the users are not shown means that they are not registered - chech the user's settings in sip.conf and on the sip client.

ivan (support at asteriskguru dot com)
20 September 2005 10:36:04
Jon,
If you want to transfer a call you need the PRO version of x-lite. Have in mind that the SIP protocol has problems with transfering sometimes.

Mike (mikemck at hotmail dot com)
19 September 2005 08:04:34
I've got 2 xten lite sessions loaded on different PC. When I try to dial either phone I get "call failed: 403 forbidden". Any ideas? I haven't signed up for a service yet so I'm just trying to get extensions to call each other.

Jon Eskdale (eskdale at talk21 dot com)
16 September 2005 19:51:46
Is it possible to transfer a call with X-lite and asterisk?

Thanks

maria (maria at lovelife dot org dot za)
14 September 2005 09:27:47
how do i delete dialed numbers from my exten and all the received

Marlino (marl at ioneresources dot com)
25 August 2005 09:13:35
thanks

John Senay (jsenay at cylogistics dot com)
23 August 2005 16:12:21
Can anybody tell me how to get the Linux Version of X-ten to work with FC-3 on the audio side?

SJphone works with this set up but Xten-Lite gives me the bad audio form
/dev/dsp.

lsof /dev/dsp shows that the device is not open when x-lite is started

Thanks

COMaction (info at comaction dot de)
19 July 2005 10:20:40
Supi Supi Supi, thanks fpr help.

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