SIP Service


This is the home of the well-known free 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?
Please report problems with the SIP service to email address sipservice(at)

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
SIP proxy:
empty or
NATi/STUN settings:
Not required (leave empty)

As fully featured cross-platform (Win/Linux/Mac) free software softclient 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 at and use it on Windows, Mac OS X, and Linux.

To configure your account click on the "File" menu and select "New Account". In the "New Account" dialogi select the "" 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 ->"

Username:      SerWEB login
Password:      SerWEB password

snom settings


Identity active:
Displayname:   Your name
SerWeb login
Password: SerWeb password
Outbound proxy:

Authentication username:
 SerWeb login

Twinkle settings

User, SIP account:

Your Name:
Your name
User name*:    SerWeb login
Organization: Your organization

SIP authentication:

Authentication name:    SerWeb login
SerWeb password

About NAT settings

There is a STUN server running on or For the 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 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:

  • 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  ( or voicebox (

To record your personal greeting prompt, dial 1001  (

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 PSTNi - identity change

With the service only pure VoIPi calls can be made; does not sell minutes into PSTN. To call a normal phone number from, 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 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: (use as domain), sipgate, (use 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 will provide iNum numbers, so users will be reachable from the PSTN directly.

Peering prefixes to other domains

The SIP service allows inbound and outbound calls from and to other domains. Just use the full address including the domain, for example

For phones that don't support entering alphanumeric addresses, and for convenience, there is some peering number prefixes:

Domain  Prefix 2220 2222 2224 2225 2223
 US tollfree (callwithus) 2227
 US tollfree (alcazarnetworks) 2228 (2229888 for the users conf) 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

Echo test call

Call echo ( 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 ( to listen to a wonderful fado of Anamar. Call early_music ( to listen to this as RBT (in an early media dialogue).

Have-my-domain! - host your own SIP domain at

With this feature you can register your own domain to be hosted on SIP server. If you would like to have your domain hosted on 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 and port 5060.

Register your domain at have-my-domain  

When you have registered your domain on, you can simply use your domain name instead of For example, if you have set the SRV record for SIP in DNS of your domain to, and registered at the link above, you can create the user with your admin user, 
and then use the username myuser and domain in your SIP phone.

After you have registered your domain at the link above, you can change the SRV entry  to port 5060 to be fully RFC compliant ( is a CNAME).

Note: If your phone does not honor SRV records, you might need to set as proxy in the phone settings.


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