PBX connect is the main extension. It is a collective account that does not contain any telephone numbers at first. Only after a successful registration for the PBX connect you can register the desired telephone numbers (PBX number) through your user account.
You can register an unlimited amount of telephone numbers. The monthly fee is € 1.50 per telephone number.
You can make up to 10 telephone calls per telephone number at the same time.
No, you can register only landline numbers on which you own the right of use.
It is necessary to keep the telephone connections and numbers at your current provider. VOIPGATEWAY can only be used for outgoing calls to landline and mobile numbers.
No, we do not offer full porting of phone numbers. This is why you can use VOIPGATEWAY only for outgoing phone calls (except for the use within the VOIPGATEWAY net).
No, VOIPGATEWAY does not offer new telephone numbers. You can register your existing telephone numbers at VOIPGATEWAY and make phone calls with these numbers through VOIPGATEWAY.
Requests on the CLIP-adjustments can be communicated to our customer service. We necessarily need a confirmation that the customer is the rightful owner of this telephone number or has the right of use for it.
The invoicing will be accomplished with each passing minute, which means that every commenced minute counts as a whole minute.
Yes, all telephone calls within the VOIPGATEWAY net are free. If you have a VOIPGATEWAY connection at different company locations, you can make telephone calls within the locations at no charge, independent of the location.
No, VOIPGATEWAY has only one uniform tariff.
The telephone system can be configured so that all calls on service numbers are routed via your landline automatically.
No, VOIPGATEWAY can only be used in the prepaid mode. After the registration, you will receive an invoice via e-mail with the installation costs, the subscription fees for the PBX connect connection for the first 12 months and a talk credit of € 50. Afterwards, you can charge your account with a freely selectable credit.
For reasons of safety, the first invoice must be paid by bank transfer. Subsequently, the option of payment by credit card in your account will be activated and you can charge your VOIPGATEWAY account with your credit card.
You can generate and print receipts of payment at any time in your user account.
You can find and download all conversation reports and cost positions in the desired time in your user account under "xDR Browser".
As soon as your balance has reached € 20, you will receive an automatic e-mail notification. Afterwards, you need to recharge your account with credit. If you do not execute any payment and the balance on your VOIPGATEWAY account is reduced to € 0.-, the service will be interrupted automatically.
At the moment, payments can be made with Mastercard and Visa.
You can click at the top of the website on „Account Login“ or directly enter https://my.voipgateway.com in the browser.
If you have registered for a PBX connect connection, you will receive a confirmation e-mail within 24 hours (on working days) with the corresponding access data.
You can click on „Passwort-Wiederherstellung“ at any time and you will receive an e-mail with the new password. As soon as you are logged in, you can change your password again.
In your personal user account, you can manage address details, register new telephone numbers, make credit card payments, create receipts of payment, check conversation and cost overviews and much more. You can find a more detailed instruction for the VOIPGATWAY user account here.
You do not receive access to the web interface with a trial account. To test our VoIP services, you merely need the user data included in the e-mail, which need to be configured on the telephone system or the SIP-terminal.
No, you do absolutely not enter into any commitment with a trial account and you can test it for 3 months at no charge (including € 50.- starting assets).
No, only one trial account can be opened per enterprise/person.
You cannot recharge the trial account yourself. As a rule, € 50.- will suffice for first tests of our telephony services. If you require more credit for your testing, contact our customer service via the contact form.
The trial account will be deactivated automatically after 3 months. Any possible remaining assets on the trial account is forfeited (no payment or transfer to another account).
You should count with 100 kbit/s per conversation (up- and download). If you have 10 conversations at the same time, a bandwidth of 1 Mbit/s up- and downloads is required.
Thanks to the existing analogue/ISDN/PRI-connection at your landline provider, the telephone system automatically switches from SIP to landline (provided that the telephone system is configured correctly).
No, you cannot make phone calls on emergency numbers with VOIPGATEWAY. Emergency calls have to be done via your landline connection. However, the routing on your telephone system can be configured so that in case of an emergency call, the phone call gets automatically routed via the landline.
If you hear the message"the password is invalid" after the connection establishment, you have configured the wrong VoIP-password on your telephone system. Please use the VoIP-password that you received from VOIPGATEWAY via e-mail.
For reasons of safety, the VoIP-password cannot be changed by the customer itself.
If you receive this message, you have entered your user identification in the configuration incorrectly. Your user identification is always the same as your telephone number including the country code and must necessarily be written out in the format as well as in the confirmation e-mail.
The error message"This call is not covered by your tarif plan" can have two reasons.
1) You do not have enough credit on your account to make this phone call. In that case, you need to recharge your account with credit.
2) You have dialled the phone number in the wrong format. The canonical format with + is not supported.
Problems with the voice quality can have several reasons:
a) If you have dropouts, the latency time of your internet connection is too big. It is possible that the bandwidth of your internet connection is too small. If this problem occurs too often, you need to contact your internet service provider.
b) A too strongly compressed audio codec is mostly the reason for distortions (hollow voice). Please try it with another audio codec e.g. G711a/G711u (PCMA/PCMU). You can define this in the configuration of your telephone system.
c) In most of the telephone systems, you can eliminate echoes with the setting "Echo Cancellation".
d) If the line remains mute or the voice communication is one-sided: In this case, a firewall upstream or a router is probably blocking the audio ports needed. Please check your network settings.
The following port is important for the VoIP communication: 5060
UDP: On that port registrations and call signalling take place.
RTP port range:
In this port range voice data is transmitted. This port range can be configured in your telephone system. The setting is usually called “Audio port range” or “RTP port range”. If you do not want to use the standard port range, it can be set in any manner. Make sure that the port range is higher than 10.000, so that no ports of services are exceeded and 2 ports per active call are available. If for example 10 phones are available, supposing that every phone has a three-party conference, the port range should contain at least 60 ports.
The port 5060 UDP and the RTP port range has to be enabled on the firewall.
With the port 5060, we recommend to leave, if possible, a source restriction. The port 5060 needs to be open for the IP range 18.104.22.168 – 254. This IP range is reserved for the VOIP server by us. There must not be any restriction defined for the RTP port range. The voice data is mostly transmitted peer-to-peer. This port range is not safety-critical because the telephone system only receives language packs on these ports.
Incoming packs have to be forwarded to port 5060 from the router to the telephone system, so that the telephone system is also available for incoming calls.
There are two possibilities to achieve this:
1. Activate “NAT Keep Alive” on the telephone system. Thereby, a pack on the port 5060 will be sent every 20–30 seconds to the sipcall server. Thus, the entry in the dynamic NAT table is kept.
2. Install the port forwarding for the port 5060 on the telephone system. Afterwards, all packs, which arrive at port 5060 will be forwarded to the telephone system; SIP requests from other IP's than from the sipcall servers included. This is the reason why this option should only be used in combination with a source restriction on the firewall.
There are a lot of routers on which SIP-ALG does not work SIP-conformant. SIP-ALG can possibly be the reason for problems with the voice transmission. During the call connection, the IP address of the gateway, respectively the IP address of the calling VoIP connection, is given directly as the destination address for language data to maintain optimal voice quality. Some SIP-ALG's substitute these IP addresses simply with the SIP server address, whereby the language data are sent to the wrong IP and do not arrive. With such problems the SIP-ALG needs to be deactivated on the router. This setting, depending on the router, is named as „SIP processing“, „SIP-ALG“ or „SIP Application Layer Gateway“.
1. Do not use any standard passwords and user names:
Avoid standard passwords and standard user names, which you have configured for your telephone system/PBX or telephone. Standard users on the telephone system/PBX often begin at ’10’ or ’100’. As a user you can for example choose ’5616725324’ and define a secure password (e.g. d1k39!&12kl).
Note: Unfortunately, not all telephone systems/telephones support special characters in the passwords and at some telephone systems, the user names are given and cannot be changed.
The passwords and user names described above are not to be mistaken with the VoIP-password and the user name of VOIPGATWAY.
2. Firewall (port 5060):
At some firewall devices, the port 5060 can only be opened for the VOIPGATEWAY server and directly enter the DNS name of the gateways. Direct entry of IP addresses is not recommended. If it is only possible to enter IP addresses, we recommend to create wild cards for the following 2 networks: 212.117.203.*. For detailed information please contact the firewall manufacturer.
3. Call connection only from the local network:
Some telephone systems/PBX have a safety function that only allows one call connection from the local network while in activated condition. For detailed information please contact the device manufacturer.
Backbone Solutions AG offers a wide range of VoIP-services with the platform www.sipcall.ch especially for Swiss enterprises with location in Switzerland.
SIP/RTP is transferred unencrypted on the internet as standard (that applies to VOIPGATWAY, too). The passwords are transferred encryptedly as hash. It is very time-consuming to intercept conversations on the internet (it is very easy to do so with local common telephone systems) because the single voice packages often use other ways. SIP or RTP provides an encryption with TLS/SRTP. However, this technology must be supported by the terminal/telephone system.