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3. Configuring IP Phones for use with Asterisk

Overview:
Asterisk allows people to communicate using the internet. It is one of the cheapest ways to talk. The PC clients connect to each other through an Asterisk server which uses a Linux/Unix OS.

Asterisk calls can be passed through different channel protocols. The most popular forms of calls are H323, SIP, IAX or Zap. Depending on the type of channel you use, the user must have the appropriate software to handle the call type.

Example:
If are a registered on an Asterisk PBX(or other PBX) as a SIP user, you are required to use a SIP phone client such as Idefisk 2.0 or SIPPS. Or, if you are registered as an IAX user you must use an IAX phone client such as iaxcomm.


SIP/IAX Client Configuration:
Now, we will describe how you can register SIP and IAX users. This includes the adjustment of all Asterisk and client phone settings.

First, make sure Asterisk is installed. Most of the files to be modified are stored in /etc/asterisk.

To make calls with this kind of telephone system you should have the following:

-Connectivity to a Asterisk PBX for call handling
-Users using PCs with SoftPhones for calling internally and externally


All configuration information for the Asterisk PBX knows is written to the conf files located in /etc/asterisk.

Example:
All information for SIP users is stored in sip.conf and for IAX users in iax.conf. Numbers are read from extensions.conf. Voicemails are stored in voicemail.conf etc.


To place a call through Asterisk, registration to Asterisk is not required. However, you do need a username and password which will allow you to call someone who is registered.

To receive a call, you must be a registered user and have an extension.

To Register SIP Users:

All users must be registered in sip.conf and all valid extensions must be declared in extensions.conf

Follow these steps for:

-Registering a user in sip.conf
-Registering a number in extensions.conf


1. Go to the /etc/asterisk directory on your Asterisk server. Open sip.conf and register the following test user.

Note that ALL characters after ; (semi-colomn) are interpreted by Asterisk as a comment.
[ivan]
This means we are registering user 'ivan'

type=friend
This means the user can place or receive calls. For INBOUND calls only, use peer as type. For outbound calls only use user as type.

username=ivan
This declares that our user will be named 'ivan'

secret=pwd_ivan
This creates the password for the user to login/authenticate on Asterisk

host=dynamic
This sets dynamic IP for the host. You may also define this as a static IP

context=tutorial
This defines the dial context for the user which in this case is tutorial. In Asterisk, outgoing numbers are divided in groups called contexts in order to separate/define different needs for different user types. For example, a context for local calls, another for within the city, and another for international calls and so on.


This completes the user registration process on Asterisk. Now let us register an extension for the user.

2. Go to /etc/asterisk and open extensions.conf


[tutorial]
This registers the context tutorial

exten => 1234,1,Dial(SIP/ivan)
when dialing number 1234, Asterisk will first Dial the user xlite through SIP protocol. The command is : exten => number, priority, Dial(protocol/user). The priority determines the sequence in which the extensions will be executed.


We now have a registered SIP user and extension on Asterisk. Follow this same process to register another SIP user and extension in order to place test calls.

Start by registering the second user in the same way in /etc/asterisk/sip.conf



Register the extension(4321) in /etc/asterisk/extensions.conf in the same context = tutorial.



Now when user 'ivan' or any other user from the tutorial context dials 4321, the user 'test' will be called.

3. The final step is to register the user to a compatible softphone. However, softphones will be reviewed later. For now, just make sure you have registered the users and extensions.

More detailed configuration information for a series of phones can be found here:

01. asterisk with softphones
02. asterisk with hardphones

To Register IAX Users:


IAX (Inter Asterisk Exchange) is another imortant protocol supported by Asterisk. To use this type of communication the users must be registered in /etc/asterisk/iax.conf.

Follow these steps for:

-Registering a user in iax.conf
-Registering a number in extensions.conf


To Register a user in iax.conf:

Create user 'ivan_iax' with the same username and join it to the tutorial context. Set the host IP to dynamic and create a password as described previously.



The user is 'ivan_iax' and type is 'friend' again (Inbound/Oubound calls allowed). Secret equals your chosen password, host equals 'dynamic IP' and context is 'tutorial'.

Now register a second IAX user following the same steps.



To call the first user 'ivan_iax' dial 1111. Note that the Dial command when using IAX2 protocol is : Dial(IAX2/username@username)



To call the second registered user, create extension 2222.



You have completed the registration of 4 users(2 SIP/2 IAX) and 4 extensions. Now, you must reload Asterisk. When any change is made in conf files from /etc/asterisk/ or changes that relate with some of these files, you must type 'reload' in the Asterisk Command Line Interface (CLI) to make the changes effective.

The users and extensions are now registered on Asterisk but the users must also be registered on a SIP or IAX client softphone. As stated, we will review softphones in a later in the tutorial.


More detailed configuration information for a series of phones can be found here:

01. asterisk with softphones

02. asterisk with hardphones

 
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https://www.voipbusiness.com/benefits-of-voip/
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How do we cofigure the asterisk PBX username "+254XXXXX@domain" to register with IMS core as opposed to "254XXXXX".
shahin (shahin_4s at hotmail dot com)
10 February 2015 13:35:18
Hi,
I am trying to connect two x lite clients on vista machines with an asterisk server on linux. However, my x lite is not able to register with the server , "database show" does not show their entries.. Please help me with this

Following are my config files:

sip.config
[ivan]
type=friend
username=ivan
secret=XXXXX
host=dynamic
context=Testing
allow=all

[test]
type=friend
username=test
secret=XXXXX
host=dynamic
context=Testing
allow=all

extensions.config
[Testing]
exten => 1234,1,Dial(SIP/test)
exten => 4321,1,Dial(SIP/ivan)

pls help me
dodo (hellwitu at gmail dot com)
03 August 2009 09:16:12
HELPPPPPPPPPPPP!!!!
M goin nuts here....
i have installed fedora 9...
asterisk 1.4.21.2..
usin x-lite 2.0 (also tried wit zoiper)
my sip files and extensions are good...
the extensions work great....

i can make calls to myself from any of the registered users i have created...
but noone else can connect to my server from any other system( linux and windows)...

been breakin my head for over a week now tryin alot of stuff....
and i have only 3 days left to submit my work wit the frontend !!!!...
its for a office wit dynamic ips...

initially everyone could connect but now noone can... except me to myself...

anyone PLZZZZZ HELP ME OUT...!!!
its URGENT...
kath (kdsanmiguel at apc dot edu dot ph)
20 July 2009 09:30:22
how to disable registration when i using host=static???

sir i need help on this.....
tnxs!
aravind (battipallyaravind at gmail dot com)
04 May 2009 17:04:58
how to configure a 2 ip settings in asterisk server
ahmed  (nuur00 at hotmail dot com)
31 January 2009 19:29:11
I like to learing confeguretion ip phone and how do I use IP phone
Rucha (rucha dot lale at gmail dot com)
29 January 2009 03:41:02
Hi,
I am trying to connect two x lite clients on vista machines with an asterisk server on linux. However, my x lite is not able to register with the server , "database show" does not show their entries.. Please help me with this

Following are my config files:

sip.config
[ivan]
type=friend
username=ivan
secret=XXXXX
host=dynamic
context=Testing
allow=all

[test]
type=friend
username=test
secret=XXXXX
host=dynamic
context=Testing
allow=all

extensions.config
[Testing]
exten => 1234,1,Dial(SIP/test)
exten => 4321,1,Dial(SIP/ivan)

In the x lite -> System Settings -> SIP Proxy -> Default i have entered the following fields
Enabled = Yes
Display Name = ivan
Username = ivan
Authorization use = ivan
Password : XXXX (same as in secret field)
Domain/Realm = 192.168.4.101 (IP of Asterisk)
SIP Proxy = 192.168.4.101 (IP of Asterisk)
Out Bound Proxy = 192.168.4.101 (IP of Asterisk)
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Tanmoy (tanmoyiem at yahoo dot co dot in)
09 January 2009 13:37:46
hi
1. how to change directory in asterisk server?
2. what is actually "10.3.3.25-PuTTY"?
we entered in "/etc/asterisk"...but could not open this "10.3.3.25-PuTTY"...so how to open this?plz help...
rajesh (rajeshv18 at yahoo dot com)
17 November 2008 13:24:59
hai, i am new to asterisk. i installed asterisk software in CentOS 5.2 and installing and configure asterisk software. after that i am login to xlite software, i am getting error 408
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14 September 2008 21:02:32
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abdisalan (abdisalanomar46 at hotmail dot com)
13 June 2008 12:35:39
i am abdisalan needs to be registion with this asterisk
Venkateswararao (venkat at stellentsoft dot com)
10 June 2008 14:32:32
Hi to astrisk forums,

This is venkateswararao I am new to Asterisk I was installed asterisk software aand I configure ASterisk I menstion Bellow

[Stellent]
type=friend
username=Stellent
secret=Stellent
host=dynamic
context=tutorial

Am i trying to make a call With sip phone Xlite I got Error $09 Call faoiled Please tell me how to rectify this problem I am using Gafachi account to make calls from x-lite Sip phone How Can i Rectify this problam?
With respectable Regards,
Venkat
bacamhou (wasaza88 at yahoo dot fr)
28 May 2008 14:46:34
how to make the sip in the x lite and how to get credit in the x lite? thanks
Wil (wilson dot upindi at standardbank dot co dot mw)
24 May 2008 12:49:28
I have set the server and the client as above but am getting call forbedden on my xlite
notrelevent (junk at akesons dot com)
02 May 2008 19:04:02
Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Really destroying SIP dialog '45d3b1743b95838964d7b68f38dcbdf4@127.0.0.1' Method: INVITE

This usually indicates no return route from the destination. IE: if you are able to get out your firewall to connect to your provider, there is no rule to allow the traffic back. I say usually!
Shahanshah (shahanshahs at gmail dot com)
11 March 2008 08:47:44
best site for Asterisk.
faheem (faheemahmed007 at gmail dot com)
21 February 2008 19:09:32
hi

1)is there any way that we can block the sites of the agents.
2)how can we safe asterisk.
faheem (faheemahmed007 at gmail dot com)
21 February 2008 18:58:34
hi

1)is there any way that we can block the sites of the agents.
2)how can we safe asterisk.
Styop (stepan87 at gmail dot com)
13 February 2008 20:59:27
Hi

Can someone give me example file for H323
h323.conf & extensions.conf please

Thanks.
souvik (sadhu dot souvik at gmail dot com)
12 February 2008 17:20:33
Hi,
I m facing problem to register xlite soft phone with my asterisk server.
I have already configured sip.conf and extensions.conf

both r ok......
and the configuration shown in this tutorial which shows how to configure xlite client , that is too old version.
can anybody update with the new one...and help me by placing answer.
"I set up Asterisk and i try using SIP channel but I did not susscess
once i use soft phone Xlite it has massage "Time out 408ms".
I'm beginer in use Asterisk"..
thanks in advance
souvik
souvik (sadhu dot souvik at gmail dot com)
12 February 2008 17:20:11
Hi,
I m facing problem to register xlite soft phone with my asterisk server.
I have already configured sip.conf and extensions.conf

both r ok......
and the configuration shown in this tutorial which shows how to configure xlite client , that is too old version.
can anybody update with the new one...and help me by placing answer.
"I set up Asterisk and i try using SIP channel but I did not susscess
once i use soft phone Xlite it has massage "Time out 408ms".
I'm beginer in use Asterisk"..
thanks in advance
souvik
wale (mails2wale at yahoo dot com)
15 January 2008 09:06:00
help
my GXW-4004 would not register when i configured it.please can someone help me.
thanks
goga (gurjet_sh at yahoo dot com)
26 November 2007 14:05:22
I want to add Message, when line is bussy

PLEASE STAY ON THE LINE, ONE OF OUR TECHNICAL EXECUTIVES WILL ATTEND YOUR CALL SOON OR YOU MAY CALL LATER. SORRY FOR THE INCONVENIENCE.

Tell me how to do that in perticular No.
hubert (hubert dot he at gmail dot com)
10 October 2007 10:41:13
how to configure the asterisk server with BLF function and how to add BLF function in the extensions.conf file
anmol malik (anmol dot malik2007 at gmail dot com)
30 July 2007 11:48:41
i wanna free phones
Hiten (hiten_be2005 at yahoo dot co dot in)
28 June 2007 12:57:57
I get this "Login Time out! Contact Network Adminstrator" Asterisk is on a Linux box.

My settings are
"sip.conf"

[ivan]
type=friend
username=ivan
secret=123
host=dynamic
context=tutorial

[test]
type=friend
username=test
secret=456
host=dynamic
context=tutorial

"extensions.conf"
[tutorial]
exten => 1234,1,Dial(SIP/ivan)
exten => 4321,1,Dial(SIP/test)

I have Xlite softphones , one on windows box and another on Linux ( where Asterisk is installed ).

Xlite on Linux throws me error "Login Time out! Contact Network Adminstrator" and Xlite on windows gives me " Registraion error 408

Please help. Thanks.
hilton (hilton dot bloomberg at gmail dot com)
13 June 2007 09:50:05
I really need help, i have no experience with linux or asterisk, i need a "idiots" guide does anyone know where i could find one on the net.?
deepak (deepak_onlyenjoy4u at yahoo dot co dot in)
07 June 2007 15:38:16
great,can u pls provide details on H.323 Protocol
ishaq (x_657 at hotmail dot com)
05 June 2007 08:43:33
hi.............i have succesfully installed asterisk on linux.now i would like to register xlite phone with it sio i need the appropriate guidance in regards to configuring sip.conf and extensions.conf.i will be thankful to u.
mahad (x_gt2003 at yahoo dot com)
25 May 2007 19:43:05
i want free phone
Oge (loveoge at hotmail dot com)
22 May 2007 15:38:05
I am really new to asterisk. It is virtually new in my country so getting help is a little restricted. I have been trying to configure a hard phone with asterisk but i keep getting this message

TFnetLinux:/etc/asterisk # [May 22 14:27:50] NOTICE[4137]: chan_sip.c:14606 handle_request_register: Registration from '"300" <sip:300@86.62.2.155>' failed for '86.62.2.157' - No matching peer found

Pls i need help and very soon

thank you
pratik (kotkar dot pratik at gmail dot com)
21 May 2007 04:33:37
Had a doubt...

Im using Asterisk and since I dont have access to 2 PCs for the testing, Im suing Virtual MAchine(VM)...My virtual OS is Red hat linux...while the host is Win XP...

My asterisk is running on the virtual linux OS...But im only able to connect to it using a softphone on the virtual OS...but not thru the host...Does this problem occur??? or is it some config problem?? the config im using is same as those outlined above...

Let me know soon its urgent..
abdul (abdul_rauf_52 at yahoo dot co dot in)
24 April 2007 12:51:16
hi,
i am abbul plz halp me i want this systam options pls haertly help me ok bye thank you
Ramesh (mesha123 at rediffmail dot com)
17 April 2007 14:52:16
Hi, I have installed asterisk on fedora already. But I dont know how to install the SJ Phone on linux platform. Tell me the steps for installing SJ Phone and explain how to use it.
Greg (greg dot barr at bytecafe dot net)
12 April 2007 23:13:14
I have a SPA-841. I want to assign 2 different extensions in extensions.conf to the two extensions on the phone. I have tried to change the port on the second ext, but when dialing the extension I set up for it, asterisk says Everyone is busy/congested at this time. When I dial from either ext on the phone, asterisk sees it as coming from ext 1. Can you have to entries in sip.conf for the same phone with different parameters?
sridhar (sridhar dot siricilla at gmail dot com)
28 March 2007 15:13:03
i got following error while installing the x-lite in the Linux
[root@DT4397-sridhar xten-xlite]# ./xtensoftphone
I/O warning : failed to load external entity "/root/.Xscrc"
rani (ranitrue at gmail dot com)
14 March 2007 10:49:06
Hi,
I am new to asterisk. I would like to register sjphone with asterisk server. I have installed Asterisk server. After creating a new profile and necessary configurations in sjphone i am getting the result as not registered.
IvanJacobs (ivan at dextrous dot co dot za)
13 March 2007 15:30:55
hi guys, i\'m pulling out my hair here, need some help pls, i\'m all new to this asterisk thing, and i\'m trying to set up 2 sip servers to talk to each ohter via the internet, so for eg, dialing 5xxx woud dial sip 2, and dialing 6xxx would dial sip 1,


pls muchly, and mostsest,
Aditya (adityaakumar at hotmail dot com)
08 March 2007 09:43:08
Hi,
I have successfully register the X-lite soft phone with Asterisk server,But unable to establish a call with other extension,also got 503 response (service unavailable)
The Asterisk verision is:asterisk-1.2.15
Whenever I am calling to other extension,On asterisk server I have found the following messages.can u tell me where I was wrong
" 8 14:09:28 NOTICE[2981]: app_dial.c:1055 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/192.168.98.47-097dc098' status is 'CHANUNAVAIL'"

Looking for advance comment

Thanks
Aditya
Talha  (talha dot faraz at hotmail dot com)
20 February 2007 06:35:24
thankx
hossameldeen (hossam_eldeen_1 at yahoo dot com)
30 January 2007 02:54:18
i.have a big porplme at my coling
manuel paz gonzalez (manuelpaz2005 at yahoo dot es)
21 January 2007 03:30:53
QUIERO MAS INFORMACION SOBRE LA CONFIGURACION DE IDEFISK YA QUE TENGO PROBLEMAS CUANDO LLAMO O SEA LA PERSONA NO ME OYE SIN EMBARGO YO SI LA OIGO A ELLA Y MI MICROFONO ESTA EN BUEN ESTADO .QUIERO OBTENER A TRAVEZ DE USTEDES UNA LISTA DE LOS CODIGO QUE DEBO USAR PARA LOS DIFERENTES PAISES Y ESTADOS .LES PIDO QUE LO MANDEN EN IDIOMA ESPAOL MUCHAS GRACIAS.
pablo (pablo at hotmail dot com)
31 December 2006 20:05:10
Yes, great work!
thanks youu
Rajeswari (rajee_2k1 at yahoo dot co dot in)
04 December 2006 10:32:13
Can you also exlain how to configure a CTI server to use with Asterisk in a call center environment
dar (daria dot morgendorffer at centrum dot cz)
16 November 2006 20:03:01
Connected to Asterisk 1.2.7.1 currently running on ubuntu 6.06 LTS:

When I try to call 2222 from 1111, I get

Registered IAX2 'test_iax' (AUTHENTICATED) at 192.168.2.183:4569
Nov 16 12:49:57 NOTICE[3773]: chan_iax2.c:5692 update_registry: Restricting registration for peer 'test_iax' to 60 seconds (requested 300)
Nov 16 12:50:12 NOTICE[3773]: chan_iax2.c:7215 socket_read: Rejected connect attempt from 192.168.2.154, request '2222@tutorial' does not exist

There is no overtype (a proof upon request)
simo med (simoelmar at gmail dot com)
13 November 2006 10:45:55
hi all!
i'm asking for how to add a user in my asterisk server for x-lite softphone!
please send how to do it!
simo
jadu (jadu_mia2010 at yahoo dot com)
26 October 2006 21:52:34
i don,t know how can i config sipura.please send me how can i config sipura.
stefan (stefan at internet dot com)
13 October 2006 18:33:43
Great Turotial. I mean the whole TuT, not just the 2nd chapter. I think some people are not able to read, befor asking their questions. I read now since 2 weeks only Asterisk tutorials, tried many distris and i never had to ask someone cause all you need is here or there ;) ask g**gl3
Thx for this clean dokumentation.
masud rana (ranabiswas2000 at yahoo dot com)
05 October 2006 09:05:00
hi, i am fresh about asterisk. i want to configure asterisk to pass call over ip. pls tell me how i can do that.
1.i want to work with billing with asterisk call as well as ivr
2.if i do not use third party billing software, then what i have to do.


thanks
rana
Faisal Khan (khansherwani at gmail dot com)
30 September 2006 03:42:27
Well i think the tutorial was very very easy and good ... thanx a lot
saki (sakimustafa at yahoo dot com)
23 September 2006 13:02:02
Excellent Tutorail. Make a lot of things easier. I can call from PC to PC in my LAN. Can any one tell how do someone make a call from WAN in my address. And how to setup sjphone for that purpose?
bharath (bharath at idssoft dot com)
15 September 2006 22:45:50
how to make a call from sip to h323 and vice-versa
Peter (peterlss at yahoo dot com)
08 September 2006 16:23:50
I get this "Login Time out! Contact Network Adminstrator" when trying to connect from PC2(test) to PC1(ivan). However from PC1 i can log in and call to myself. My server is on PC1.

My settings are
"sip.conf"
;;;;;;;;;;;;;;
;tutorial
;;;;;;;;;;;;;;
[ivan]
type=friend
username=ivan
secret=123
host=dynamic
context=tutorial

[test]
type=friend
username=test
secret=456
host=dynamic
context=tutorial

"extensions.conf"
[tutorial]
exten => 1234,1,Dial(SIP/ivan)
exten => 4321,1,Dial(SIP/test)

Any ideas what is going wrong?

Please help. Thanks.
kitti (krishnamohan1238 at gmail dot com)
17 July 2006 22:40:51
i configured asterisk as like that of u gave and i'm not able to connect to the phone.its saying that user does not found could any body help me in this asspect please i'm fresher in asterisk
Sadiq (sadiq at voicesky dot com)
08 July 2006 13:23:55
Turorail was Excillent , saved much time.
internal Extention are working fine
can someone guide how to configure Asterisk for vOIP provider and extentions can dail internationally
Erlend (erlend dot gjerde at iu dot hio dot no)
26 June 2006 11:27:28
How do I run the iax.conf-file? go to c:cygrootastersiketc and write iax.conf Windows tells me it doesn't know which program to use to execute this file. If I click on the icon in Windows explorer I get the same message. I'd be gratefull if anybody could give me some advice.
Harry (not at so dot import dot ant)
21 May 2006 12:13:25
super docu!
thx 4 this!
Rehan Muzaffar (raaz97 at yahoo dot com)
27 April 2006 11:23:50
Really it was very helpful
Lacho (support at asteriskguru dot com)
25 April 2006 11:46:41
Hi federico,

This is a problem caused by misconfiguration either in your asterisk or in the phone device you are using. If you tell me your configuration in sip.conf and in the phone device I could help you, BUT please use the forum for the purpose. There is a bigger chance somebody to spot your question and you will find the solution of your problem faster. Thanks in advance

Regards
Lacho
federico (federico dot lavolpe at gmail dot com)
24 April 2006 04:06:58
Hi to all
I am new on this, I create a extention as is mentioned in this example but the login always have failed. In the Full log file of asterisk i saw a message that said user/pass does not match, but is ok.
Sombody can tell me why?
br
fede
Sixty (sixty999 at hotmail dot com)
20 April 2006 17:30:21
........asterisk: relocation error: /usr/lib/asterisk/modules/res_odbc.so: undefined symbol: ast_load
does anyone knows how to resolve this?

it happens when i try to run asterisk -c
Pratibha (pratibha_22432002 at yahoo dot co dot in)
19 April 2006 08:24:50
hi,
I have installed asterisk in VOCAL on LINUX platform.
For a prepaid call the b2bua should send an alert message to asterisk when balance is low.
how does this b2bua sends the message and how to use timer for that.
Please send me the answer as early as possible.
Le Son Phat (sonphat_le at yahoo dot com)
14 April 2006 10:38:02
I set up Asterisk and i try using SIP channel but I did not susscess
once i use soft phone Xlite it has massage "Time out 408ms".
I'm beginer in use Asterisk.Help me !!! Thanks
lee (levu202000 at yahoo dot com)
10 April 2006 03:16:29
I set up Asterisk and i try using SIP channel but I did not susscess
once i use soft phone Xlite it has massage "Time out 408ms".
I'm beginer in use Asterisk.Help me !!! Thanks
ivan (mirko at home dot hr)
20 March 2006 11:31:56
Hey even my name is Ivan! I can just copy/paste! ;)
Dan (dan at dan dot jh)
15 March 2006 23:52:08
Many many thanks, very useful.
Lacho (support at asteriskguru dot com)
10 March 2006 15:14:36
Hi Kannans,

Please, in future time ask your questions on the forum. There the chance somebody to answer you is tremendously bigger.

As for your question - You do not need any additional hardware in order to connect your IP phones to the Asterisk server. Of course you need an Asterisk server with working ethernet card and that's it. You need additional hardware only if you want to make calls using the PSTN (traditional telephone lines)

As for the quetion how to configure the asterisk I will givve you the steps in brief
1. create acounts in iax.conf,sip.conf or mgcp.conf (depending on what kind of phone you will use)
2. make extensions in extensions.conf
3. Connect your phones to the server and start making calls

For more information please check our tutorial about the iax.conf file, extesnions.conf file and Dial application. Then If you stil have questions please ask us on the forum

Regards
Lacho
Kannans (kannanltd at hotmail dot com)
10 March 2006 10:07:53
This is very usefull documents.Anybody help me step by step configure asterisk server.I want test the IP phone .
I have one asterisk server and two windows client and X-lite softphone..What additional hardware requrired for server to communicate other windows client over IP phone.

Thanks
Raheel Hassan (raheel213 at yahoo dot com)
04 March 2006 08:40:13
I wanna to call another PC in the same Intranet by softphone having a asterisk server(i.e just like a Windows messenger).
what will be the configuration needed for extensions.conf, & sip.conf???
Raheel Hassan (raheel213 at yahoo dot com)
04 March 2006 08:24:20
I wanna to call another PC in the same Intranet by softphone having a asterisk server(i.e just like a Windows messenger).
what will be the configuration needed for extensions.conf, & sip.conf???
ROBINSON SOSA (sosa_robinson at hotmail dot com)
02 March 2006 21:42:50
HOW TO CONFIGURA A LINSYS ATA WHITH ASTERISK
Tahir Aziz (mian_tahir44 at yahoo dot com)
23 February 2006 20:18:17
Good information which i was looking for..

Thanks for your great tutorial

M. Tahir Aziz
+92-333-4221686
kerberos (daemonwww at yahoo dot com)
03 February 2006 07:45:23
hello
i have set up my client with iax2 protocol
and use idefisk softphone, i can call from pc A
and the pop up window in pc B say "accept" "reject" and "ignore"
i can hear the phone is ringing and when i click
accept i can not hear anything from pc A. but keep on ringing
i short pc A and pc B can not talk. what syntax i need to add
both iax2.conf and extensions.conf, i follow your configuration

thanks
Yann (yannjezequel at hotmail dot com)
30 January 2006 10:45:38
Hello,

Despite the tutorial is clear & simple I didn't manage to to set up properly my extension.
Using X-Lite on the same host than asterisk
When I use "localhost" (127.0.0.1) as SIP Server everything is perfect.
But when I try on IP @ then for each call it says : Error 404 Timeout.
Same effect when I try on a remote PC

Thx for your support
Yhan (yhan26 at hotmail dot com)
29 January 2006 16:48:06
Hi

I've setup my extentions, but if I record a new unavail message for my user, it will always play the defautl unvail message. What could be wrong ?
adhir datta (likhon52 at yahoo dot com)
19 December 2005 07:35:25
hi ivan,
thanks for your help.i have selected Diax software which use iax2 protocalls.but when i want to register the software then there is a line
on server screen "restricting from registration".Why did it happen.please give some information about it
polane (alan_white79 at yahoo dot com)
18 December 2005 09:06:11
hi kifak
ivan (support at asteriskguru dot com)
15 December 2005 14:04:29
Hello Adhir,

First of all you have choose softphone (ex. idefisk). Then check what protocol the selected phone uses - SIP, IAX2 or other (idefisk uses IAX2). Afterwards read the manual again - create a user in iax.conf and extensions in extensions.conf and see what happens. If you have other problems write us again.
Adhir Dutta (likhon52 at yahoo dot com)
13 December 2005 07:25:48
I wanna to call another PC in the same Intranet by softphone having a asterisk server(i.e just like a Windows messenger).
what will be the configuration needed for extensions.conf, iax.conf or sip.conf???
faisol (faisol2019 at yahoo dot com)
09 December 2005 07:47:02
i need information.pls help
i made iax.conf file but my asretisk server did not register it .why is it happening?
Cesar (cerodriguez at cnu dot gov dot ve)
17 November 2005 19:26:57
How could I register a H.323 user in *?
Ashwin k.p (kpashwin at yahoo dot com)
17 November 2005 07:39:30
I have installed Express Talk with asterisk open source PBX sucessfullf, but i want to know how to transfer a call and also how to have the conferencing
zoa (support at asteriskguru dot com)
16 November 2005 16:01:04
If you have a static host, you need to disable the registrations.
Frank (franklin dot leite at gmail dot com)
16 November 2005 13:40:18
how can i use static host? it keeps complaining about not setting host as dynamic
Lacho (support at asteriskguru dot com)
08 November 2005 17:08:07
HI cuipy,

You have to type the command "reload" on the CLI
cuipy (cuipengyu at 163 dot com)
08 November 2005 07:44:05
how to reload asterisk?
Bails (bails at westcomuk dot com)
13 October 2005 14:37:14
OK, remote works with dial(iax2/username@username) as well, my mistake, damn typos!!!!!!
Bails (bails at westcomuk dot com)
12 October 2005 19:46:33
Great Tutorial, workd fine locally, but what if the iax client phone is remote (wandering), what does the dial string look like then?

I have tried with Dial(IAX2/username@username) which works locally, but not remotely. I know that Dial(IAX2/username:password@host) should work but of course the host is dynamic so I cannot work this one out.
MikeM (mikemck at hotmail dot com)
19 September 2005 17:02:26
If I create an extension and configure X-ten lite (within the client) do I still need to make the entries shown above form a command line? I haven't signed up w/ a service yet and I'm having trouble getting the echo test to work.

I'm running the Asterisk@Home version 1.5 CentOS distro.
seasonzwt (seasonzwt at 163 dot com)
14 September 2005 05:04:46
Talked very simple but very useful,
thanks a lot!
But can you talk some esle about how to dail
the user in undo pbx,thank you
Bloodygun (bloodygun at banda dot lv)
10 September 2005 14:26:56
great tuttorial!
if your clients is behind a firewall remember to put

[ivan]
.....
nat=yes

in each user configuration
Zeshan (zeshan at datels dot com)
09 September 2005 08:43:48
In my opinion its a very good&simple tutorial ..

With regards,

Zeshan
Angie (angelicamancilla at shaw dot ca)
08 September 2005 07:31:56
How do I open the sip.conf. I just started with this tutorial, so far, I got to the right directory. Thanks
WebMaestro (asiby at hotmail dot com)
29 August 2005 07:50:23
Great!

This finally helped a lot. Great and simple explanations.

Many thanks.
Blessen (blessen at poornam dot com)
26 August 2005 21:48:58
I tried your settings. But i could get my kphone to work. Mu configuration is posted here.
http://www.voipuser.org/forum_topic_2179.html

blessen
zoa (support at asteriskguru dot com)
08 August 2005 21:41:58
No, you should not add the users from the CLI, you should add them by using sip.conf or iax.conf.

If you have more questions, please post them on the forum. ( http://www.asteriskguru.com/board/ )
Bader Dip (bader at tevisat dot net)
07 August 2005 06:23:56
A good tutorial in my opinion but I'm still no sure if should add the users from the asterisk console??
ronoc (asteriskguru at cuimalo dot e4ward dot com)
03 August 2005 11:15:10
Yes, great work!
Peter (peter at cyclix dot se)
19 June 2005 20:18:08
many thanks for a great introduction to asterisk !!
 
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