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, SERWeb and SEMS. It also uses third-party equipment, currently iptego's Security and Monitoring Application.
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'.
snom settings Login
Twinkle settings User, SIP account:
SIP authentication:
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:
To dial into the voicebox for checking your messages dial 1000 (sip:1000@iptel.org). To record your personal greeting prompt, dial 1001 (sip:1001@iptel.org). Conference calls The iptel.org webconference (non-secure link: http://webconference.iptel.org) 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. 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).
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