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).
It also uses third-party equipment, currently FRAFOS' ABC Session Border Controller.
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
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
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 calls.
In the "forward" tab of the account page in SerWeb, there are several options that control call forwarding settings. Here a SIP URIi 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:email@example.com) or voicebox (sip:firstname.lastname@example.org).
To record your personal greeting prompt, dial 1001
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 PSTNi - identity change
With the iptel.org service only pure VoIPi 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, 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:email@example.com.
For phones that don't support entering alphanumeric addresses, and for convenience, there is some peering number prefixes:
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 firstname.lastname@example.org.
Echo test call
Call echo (sip:email@example.com) 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:firstname.lastname@example.org) to listen to a wonderful fado of Anamar. Call early_music (sip:email@example.com) 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 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 firstname.lastname@example.org with your admin
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
Note: If your phone does not honor SRV records, you might need to set sip.iptel.org as proxy in the phone settings.