SIP Service

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 [1], SERWeb [2] and SEMS [3].

It also uses third-party equipment, currently iptego's Security and Monitoring Application [4].

Go to 'My account' [6]

Forgotten your password? [7]

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:	SerWEB login
Password:	SerWEB password
Domain:	iptel.org
SIP proxy:	empty or sip.iptel.org

snom settings

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

Twinkle settings

User, SIP account:

Your Name:	Your name
User name*:	SerWeb login
Domain*:	iptel.org
Organization:	Your organization

SIP authentication:

Realm:	iptel.org
Authentication name:	SerWeb login
Password:	SerWeb password

Call forwarding and Voicemail settings

By default, voicemail2email is enabled for offline users and not answered calls.

In the "forward" tab of the account page in SerWeb, there are several options that control call forwarding settings. Here a SIP URI as destination for on-busy, 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).

Conference calls

The iptel.org webconference [8] (non-secure link: http://webconference.iptel.org [9]) allows you to create and control conference rooms through a web page.

This conference server is also accessible for dial-in at sip:conference@iptel.org . Using the prefix 000777 leads you directly into the conference room, e.g. calling sip:000777123@iptel.org goes directly into the conference room 123.

A special feature of this conference service is that you can connect people not only in the iptel.org SIP service network, but also in other networks,

you just need to provide an outgoing line for it. For example, if you enter your account of your favourite VoIP provider that offers cheap PSTN termination, you can do cheap PSTN or mixed PSTN and VoIP conferences.

Calling into PSTN - identity change

With the iptel.org service only pure VoIP 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 SerWeb page as custom PSTN gateway (custom PSTN gateway user, custom PSTN gateway domain, 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). It does not work with providers that require registration to place calls.