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SIPgate - German-based VoIP Provider
VOIP to PSTN as well as PSTN to VOIP service provider (voicemail is planned and in Germany available as a test).
Subscribers to this service get a public phone number that people can call (in "selected cities" in Austria, Germany and UK). It is free to sign up and to call anybody on the Sipgate network as well as some other VOIP networks (FWD, Iptel, Sipphone.com, IAXTel, Telio.no, freenet, web.de and more). There is no monthly charge for the assigned phone number. Sipgate works with pre-paid accounts, which you can recharge by credit-card or bank-debit (min. €10-€20). There are charges for calls from VOIP to PSTN only.
Sipgate supports ENUM to resolve a phone number to e.g. a sip account. In many cities, numbers assigned by Sipgate are registered with the public DNS, so they can be resolved easily by any phone / provider / Asterisk server which is set up for ENUM. Sipgate sells a GRANDSTREAM BT-101 phone or HandyTone-486 adaptor as well as FRITZ!Box Fon pre-configured and working out of the box.
Currently, sipgate supports the followings codecs: g.711, g.729 and g.726, and unofficially also GSM and iLBC. Unfortunately, with regards to security, neither SRTP nor TLS-secured SIP (SIPS) are supported, but at least they use SIP with digest-auth.
Warning: Sipgate's proxy does not support reINVITEs. This means call transfers, call hold and anything more than the most basic call flows do not work well with this service provider.
To get SipGate working with Asterisk the parameter:
is needed, at least sometimes in the SipGate part of sip.conf, else no connection is working.
On later versions of Asterisk (1.6+ ?) "insecure=very" is not recognised and you should use the following instead:
Addon2: If you are planning to use Sipgate for outbound-calls via your * be sure to have a register => to the sipgate-server too. You only can place an outbound call via sipgate, if your * is connected to also receive incoming calls. Otherwise you will run into an error-announcement !
Addon3: Also, at the end of the register line add a "/1234567" where this number is your SipGate number, and configure an extension in your dialplan with the same number - otherwise you can't make outgoing calls and the web-presence indicator shows you as off-line even though you CAN receive incoming calls.
Addon4: To avoid authentication problems when making outgoing calls, make sure you are using xxxx@peername in your dialplan, where peername is the title of the corresponding section in your sip.conf instead of firstname.lastname@example.org. Otherwise your secret=, username=, fromdomain= and fromuser= settings won't take effect. (see Note in description of SIP channels).
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