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This is the home of the well-known free iptel.org IP Telephony service. Many
people use our services for software/hardware interoperability testing or just
as a way to call other people. The service allows incoming and outgoing calls
from/to any other IP Telephony services (note: some commercial services stop
calls to other Internet-based services).
The service is based on SIP Express Router, SERWeb and SEMS.
It also uses third-party equipment, currently FRAFOS' ABC Session Border
Controller.
Register a new account
Go to 'My account'
Forgot your password?
Mailing list, discussion and support
Bug reporting system for public SIP service
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Phone settings
Generic phone settings
The SerWeb login is the name you picked when registering. The same name
and password is used for SIP authentication. It is sent with the confirmation
email as 'Username' and 'Password'. With this account you can login above at
'SerWeb login' or 'Go to my account'.
Username:
Password:
Domain:
SIP proxy:

SerWEB login
SerWEB password
iptel.org
empty or sip.iptel.org

NAT i/STUN settings: Not required (leave empty)


As fully featured cross-platform (Win/Linux/Mac) free software softclient
iptel.org recommends Jitsi (formerly named SIP Communicator) (12-16MB
download). A small, good soft phone for windows is NCH Express talk (490K
download ), a fully featured one X-Lite . For Linux, other good free SIP
software is Twinkle , Ekiga . See also the SIP phones list .
Jitsi (SIP Communicator) settings
You
can
download
Jitsi
http://download.jitsi.org
and use it
Windows, Mac OS X, and Linux.
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To configure your iptel.org account click on the


"File" menu and select "New Account". In the
"New Account" dialog i select the "iptel.org"
option and then enter your SerWeb login as
shown in this screenshot. The new account
would then appear in your account list
(screenshot) and you can start using it
immediately.
In summary:
"New Account -> iptel.org"
Username:
Password:

SerWEB login
SerWEB password

snom settings

Login
Identity active:
on
Displayname:
Your name
Account:
SerWeb login
Password:
SerWeb password
Registrar:
iptel.org
Outbound proxy:
Authentication username: SerWeb login
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Twinkle settings

User, SIP account:


Your Name:
User name*:
Domain*:
Organization:

Your name
SerWeb login
iptel.org
Your organization

SIP authentication:
Realm:
iptel.org
Authentication name: SerWeb login
Password:
SerWeb password
About NAT settings
There is a STUN server running on stun.iptel.org or sip.iptel.org. For the
iptel.org SIP service, STUN is not required, because there is a server side
NAT traversal solution in place. In fact, it is not recommended to set and use a
STUN server, because in certain situations the server side NAT traversal that
is in place at the iptel.org SIP service is not able to detect NAT properly if
STUN is used.
Call forwarding and Voicemail settings
By default, voicemail2email is enabled for offline users and not answered
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calls.
In the "forward" tab of the account page in SerWeb, there are several options
that control call forwarding settings. Here a SIP URI i as destination for onbusy, on no answer and unconditional forwarding can be set. Alternatively you
can set the type of call forwarding to auto-attendant to use:
voicemail2email sends the voicemail as email
voicebox saves the message to your voicebox
both sends an email and saves the message to your voicebox
non disables call forwarding
To dial into the voicebox for checking your messages dial 1000
(sip:1000@iptel.org) or voicebox (sip:voicebox@iptel.org).
To record your personal greeting prompt, dial 1001 (sip:1001@iptel.org).
Conference calls
Meet-me conference calls are available with the prefix 000777. E.g., dial
000777000 for the conference room 000.
Unfortunately, the webconference service is no longer being provided.
Calling into PSTN i - identity change
With the iptel.org service only pure VoIP i calls can be made; iptel.org does not
sell minutes into PSTN. To call a normal phone number from iptel.org, you can
enter a provider account in the 'other' tab in SerWeb page as custom PSTN
gateway (custom PSTN gateway user, custom PSTN gateway domain,
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custom PSTN gateway password), and iptel.org will send all calls to numbers
starting with '+' and '00' there and authenticate the call with your credentials.
This should work with any provider that uses normal SIP authentication. It has
been tested with: nonoh.net (use sip.nonoh.net as domain), sipgate, carpo.de
(use sip.carpo.de as domain) etc. It does not work with providers that require
registration to place calls.
Calling into iNum (+883 5100)
By directly calling +833 5100 or 00833 5100 calls can be sent to iNum.
Soon iptel.org will provide iNum numbers, so iptel.org users will be reachable
from the PSTN directly.
Peering prefixes to other domains
The iptel.org SIP service allows inbound and outbound calls from and to other
domains. Just use the full address including the domain, for example
sip:auser@anotherdomain.net.
For phones that don't support entering alphanumeric addresses, and for
convenience, there is some peering number prefixes:
Domain
sipbroker.com
sipgate.net
gojiapp.com
telio.no
ekiga.net
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Prefix
2220
2222
2224
2225
2223
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US tollfree (callwithus)
US tollfree (alcazarnetworks)
conference.freeswitch.org
(2229888 for the users conf)

2227
2228
2229

For example, you can reach someone in sipgate network who has the sipgate
ID 1234567 by calling 22221234567, or you can reach the sipbroker monkeys
by calling 2220*266300. Sipbroker actually allows you to call most other sip
domains by using some mapping prefix before the star.
You can reach US toll-free numbers (1-8xxx) by prefixing the number with
2227 or 2228, e.g. dial 222718005558355 or 222818005558355 and say
'time' for a speaking clock.
If you would like to have a peering prefix configured for your or some other
domain, please contact services@iptel.org.
Echo test call
Call echo (sip:echo@iptel.org) or the vanity number 3246 for an echo test
call. You can change the buffering while in the call by pressing the star key.
Music test call
Call music (sip:music@iptel.org) to listen to a wonderful fado of Anamar.
Call early_music (sip:early_music@iptel.org) to listen to this as RBT (in an
early media dialogue).
Have-my-domain! - host your own SIP domain at iptel.org
With this feature you can register your own domain to be hosted on iptel.org
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SIP server. If you would like to have your domain hosted on iptel.org server you
first have to set DNS for your domain properly. There has to be a SRV record
for service 'SIP' and protocol 'UDP' pointing to host sip01.iptel.org and port
5060.
Register your domain at have-my-domain
When you have registered your domain on iptel.org, you can simply use your
domain name instead of iptel.org. For example, if you have set the SRV
record for SIP in DNS of your domain mydomain.net to sip01.iptel.org:5060,
and registered mydomain.net at the link above, you can create the user
myuser@mydomain.net with your admin user,
and then use the username myuser and domain mydomain.net in your SIP
phone.
After you have registered your domain at the link above, you can change the
SRV entry to sip.iptel.org port 5060 to be fully RFC compliant (sip01.iptel.org
is a CNAME).
Note: If your phone does not honor SRV records, you might need to set
sip.iptel.org as proxy in the phone settings.

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