Germany: SIP providers and connecting a number to the PBX

This page continues the general guide on connecting a provider's number to the PBX: that page covers everything that is the same for any operator — the two trunk modes, what to ask the provider, what to fill in the form, network requirements and how to test the connection. Here are the German specifics: who sells a number with a SIP trunk in the country, what that operator issues and what of it goes into Telephony → "Trunks and connections" → "Add provider".

The data was collected on 9 September 2026 from operator documentation, regulator (Bundesnetzagentur) materials and the German telecommunications act (TKG); the links are in the "Sources" section.

In brief#

• You can buy a number with a SIP trunk in Germany, and this is the best documented market we have described. Your business needs neither a licence nor a registration: the service is sold by an operator, and you are simply its subscriber. Four operators publish a complete ready-made Asterisk configuration.
• Seven operators publish their server address and mode, and all seven work in registration (REGISTER) mode: easybell, sipgate, Placetel, dus.net, fonial, peoplefone, reventix. You can connect without calling a sales manager first.
• If no operator has been chosen yet, take easybell: it publishes a ready Asterisk configuration with exactly our codecs (a-law, u-law, G.722), its tariff starts at €1.95 per month with no minimum term and no setup fee, and the format of the incoming number is switchable in the control panel. The fallbacks are sipgate trunking and Placetel.
• The main local pitfall is that the incumbent is closed to a cloud PBX. Telekom's current product (CompanyFlex) requires mandatory TLS + SRTP encryption when it runs on another provider's access, and our PBX does not support TLS for trunks — that combination will not work. The older DeutschlandLAN SIP-Trunk works over TCP only (UDP is not supported at all) and is closed to new sales.
• The second pitfall is that a landline number is tied to its local area (Ortsnetzbezug). The number is issued from the calling area where the venue is actually located, and this has to be proven; a number allocated unlawfully must be switched off by the operator. The contract itself is signed with a German entity, and none of the operators we checked describes a public procedure for foreign companies — their path leads to international providers.
• The third pitfall is showing your own number on outgoing calls. sipgate, Placetel and Outbox expect the number in the P-Preferred-Identity / P-Asserted-Identity headers, which our PBX does not send: the call goes through, but with the trunk's default number. Placetel explicitly substitutes an arbitrary number from your account in that case.

Numbers in the country#

• The country code is +49. The national number is the area code (Ortsnetzkennzahl, 2–5 digits without the leading zero) plus the subscriber number.
• Length. Since 01.07.2011 only 11-digit landline numbers are issued (counting the area code, without the leading zero). In the four areas with a two-digit code — Berlin (30), Hamburg (40), Frankfurt (69), Munich (89) — connections with a single number get 10-digit ones. Shorter numbers are left over from pre-reform times.
• Extending a number (Durchwahl) is allowed: digits may be appended to the issued number, and per the ITU recommendation the total length goes up to 13 digits. ⚠️ The subscriber holds no rights to those appended numbers, and they are not ported when switching operators.
• Mobile numbers are in the 15x, 16x, 17x ranges (without the leading zero). The main service ranges are: 800 — free for the caller, 900 — premium, 180 — shared cost, 700 — personal.
• Emergency numbers are 112 and 110, dialled without an area code and free of charge.

In the "Lines" tab the number may be entered in any form: the PBX matches the last seven digits. ⚠️ In Germany seven digits are the tail of the subscriber part without the area code, so numbers from different areas may share the same tail. A Durchwahl block (…-00……-99), on the contrary, always differs. If your numbers come from different cities, check the routing of each line manually — "Add a number and routing".

Who issues a number with a SIP trunk#

The first seven rows are operators whose trunk parameters are in public documentation: their cards are below. The other German operators hand out the server address, login and password only after a contract — they have a section of their own.

easybell#

A Berlin operator for small business: SIP trunk, cloud PBX, internet access. Ordered online, with no setup fee and no minimum term (cancellation 7 days before the end of the month). Tariffs: Business 2 — 2 channels and 2 virtual Durchwahl numbers, the first 6 months free, then €1.95 per month; Business 10 — 10 channels, €4.95 per month; Business 50 — 50 channels and 100 Durchwahl numbers, €19.95 per month. A block of 10 numbers costs €19.95 once, a block of 100 costs €49.95. Number blocks for all areas, SIP DDI and CLIP no screening are included in the tariff.

What to enter in the admin panel

What to set up at the provider: order a SIP Trunk tariff → get a landline number and, if needed, a Durchwahl block → in "Rufnummernverwaltung" set the incoming number signalling format to "E.164" (then the line is matched by the last seven digits unambiguously) → route the numbers to the trunk → enable CLIP no screening if needed.

Pitfalls ⚠️ The operator has two similar addresses: voip.easybell.de is for the trunk and pbx.easybell.de is only for its own cloud PBX; they are easy to mix up, and then there will be no registration. An outdated article for Asterisk 18.9 with the address sip.easybell.de is still around — take the address from the current instruction for Asterisk 22. The number of simultaneous calls is set by the tariff (2/10/50 channels): "calls stop going through after the third one" is a limit, not a fault. The platform codecs are a-law, u-law and G.722, so our whole set passes. Source: Asterisk instruction, format of incoming numbers, SIP Trunks tariffs — 09.09.2026.

sipgate trunking#

A Düsseldorf operator, one of the oldest German VoIP brands. The product for a third-party PBX is called sipgate trunking; the operator has a public guide written specifically for Asterisk. An incoming call arrives with the full number in E.164 — the most convenient option for us: the line is matched by number without reservations.

What to enter in the admin panel

What to set up at the provider: create a sipgate trunking account → choose a number or a block of numbers → take the SIP ID and SIP password from the "Zugangsdaten" section → route the numbers to the trunk.

Pitfalls ⚠️ The operator reads the outgoing number only from the P-Preferred-Identity header, while our PBX puts the number into the ordinary From field: the call goes through, but the called party sees the trunk's default number rather than the one set in the "Show as" rule — "Outgoing calls". Hiding the number is also done with a separate header at sipgate. The trunk codecs are G.722, a-law, u-law and others, so our set passes. Source: configuring Asterisk for sipgate trunking — 09.09.2026.

Placetel#

A Cologne operator, part of Cisco (the "Webex für Placetel" brand). Besides the cloud PBX it sells a separate SIP-Trunking product for your own station and publishes a PJSIP configuration. The trunk account is called a VoIP-Ziel — it provides both the login and the password.

What to enter in the admin panel

What to set up at the provider: enable SIP-Trunking → create a VoIP-Ziel → attach numbers or a block to it → check that the numbers you intend to show on outgoing calls really belong to the account.

Pitfalls ⚠️ The operator expects the VoIP-Ziel identifier in the From field, while our PBX puts the line number there. Placetel honestly warns that in this case it will substitute an arbitrary number from your account: the call goes through, but with the wrong number. The format of the incoming number depends on the product — in the Asterisk example the account login arrives, while the checklist for hardware PBXs promises the international format +49221123456; clarify when connecting what exactly arrives on your trunk. Call forwarding by SIP means (Diversion) is not supported — build transfers with PBX rules. The codecs are a-law and u-law, that is ours. Source: connection via PJSIP, PBX connection checklist — 09.09.2026.

dus.net#

A Düsseldorf operator with the most accurate public table of network parameters in the country: addresses, ports, media server ranges and service numbers for self-testing. It has a dedicated "SIP-Trunk mit DDI" product — with a Durchwahl block.

What to enter in the admin panel

What to set up at the provider: order a SIP trunk with DDI → take the account details in the Kundencenter → route the numbers to the trunk → verify the connection with the operator's own service numbers: the echo test 00038799999 and the tone dialling test 00038799998.

Pitfalls ⚠️ Media (audio) arrives not from the address of the SIP server but from separate platform nodes (83.125.8.156–83.125.8.160) — this does not affect a connection through the cloud, but if the operator asks "which addresses do you call from", that is the answer. The codecs are not listed on the parameters page: check the audio with the very first test call, and if there is silence, ask support to enable G.711 ("Network and compatibility"). Tone dialling is transmitted per RFC 2833 — the same as ours. Source: SIP information & settings — 09.09.2026.

fonial#

A Cologne brand technically running on the Plusnet platform: the addresses in its own instruction are sip.plusnet.de. The product for a third-party PBX is called "fonial Trunking" and the instruction is written for Asterisk.

What to enter in the admin panel

What to set up at the provider: order fonial Trunking → take the SIP-Benutzer and the password in the control panel → route the numbers to the trunk → set the presentation number for outgoing calls.

Pitfalls ⚠️ The operator has a separate outbound proxy (proxy01.sip.plusnet.de) alongside the main address sip.plusnet.de, while our form has a single field: enter the main address and make sure from the status line that the registration went through. An incoming call arrives with the trunk login instead of the number — the call is accepted and goes into routing, but numbers cannot be told apart by lines. Their instruction was written for an old Asterisk version and has not been updated since: the codecs in it are u-law and a-law (ours), and tone dialling is RFC 2833. Source: fonial Trunking mit Asterisk — 09.09.2026.

peoplefone#

A Swiss group with a full German setup: a German server address and a German data centre. The peoplefone SIP TRUNK product is aimed directly at third-party PBXs — Asterisk is listed among the supported systems. The operator itself stresses that only three values are needed to connect: the address, the login and the password.

What to enter in the admin panel

What to set up at the provider: order a SIP TRUNK → ask for the line type "SIP-TRUNK" to be enabled (Kundenkonto → SIP Linien → Linien Typ; enabling it is free, but it is done by a partner or by the operator) → attach the numbers → if needed, send the signed "CLI Open" form so that your own numbers can be shown on outgoing calls.

Pitfalls ⚠️ Until the line is switched to the "SIP-TRUNK" type, incoming calls arrive "as to a phone" and the PBX will not sort them out by Durchwahl — that is the first thing to check when connecting. The group's documentation is written with Swiss examples (+41…), so verify the German formats separately. An expired test number in the CLI Open list makes all calls go out anonymously. Per a PBX vendor's test report the trunk negotiates G.711 a-law and G.722 — our codecs. Source: peoplefone SIP TRUNK — 09.09.2026.

reventix (sipbase)#

A Berlin operator trading under two brands: reventix and sipbase (the control panel is login.sipbase.de). A SIP trunk with landline numbers and blocks, free porting, CLIP no screening and a 30-day trial.

What to enter in the admin panel

What to set up at the provider: open the required connection (SIP-Trunk) in the control panel → set the area code (Ortsvorwahl) and the outgoing number (CLI) → enable CLIP no screening if you need to show other numbers of yours → route the numbers to the trunk.

Pitfalls ⚠️ The three leading zeros in the login are not a typo: the operator writes explicitly "pad the user name on the left with three zeros", and without them there will be no registration. The operator passes the called number in a separate P-Called-Party-ID header, while our PBX reads the call address — if the lines are not told apart by number, check the format on a live call and ask support about passing the number in the call address. The brand has two names and two domains (reventix and sipbase), which confuses correspondence with support. Tone dialling is RFC 2833, the same as ours. Source: reventix wiki: connecting equipment to a SIP trunk, the SIP-Trunk product — 09.09.2026.

Contract-only operators: Telekom, Vodafone and others#

These operators do have a SIP trunk, but the server address, login and password are issued only after a contract — there can be no "what to enter" table here. Ask the manager for the parameters using the list from "What to ask the provider" and enter them into the same form as for the others.

Deutsche Telekom#

The incumbent has two products, and both are problematic for a PBX in the cloud.

• CompanyFlex (SIP-Trunk) is the current product. If the trunk runs on another provider's access (and our PBX in the cloud is exactly that case), the operator requires mandatory encryption of signalling and calls and states plainly that unencrypted operation is impossible. Our PBX does not support TLS and SRTP for trunks, so the combination will not work — "Network and compatibility".
• DeutschlandLAN SIP-Trunk (including the "Pure" variant without Telekom access) is the older product, closed to new sales but working under existing contracts. Per its technical documentation: signalling is TCP only, UDP is not supported at all; the codecs are G.722 and G.711 a-law (ours); the connection address is a separate outbound proxy (reg.sip-trunk.telekom.de in registration mode, stat.sip-trunk.telekom.de in IP mode), which does not match the sip-trunk.telekom.de domain in the call headers and is resolved only through NAPTR/SRV service DNS records. If you already have such a trunk, switch the transport to TCP when connecting and check the link with the "Test connection" wizard: forget about TCP and you get "silence without errors". The IP mode ("Static Mode") is configured only by Telekom's service desk, which needs the static address of the PBX from the "Server" tab.

Vodafone Deutschland#

The product for a PBX is IP Anlagen-Anschluss / SIP Trunk. It is the only large German operator working with IP authorisation: no registration is needed, the operator is told the static address of your PBX instead. Its server address is sent to the customer by letter; the operator has no public technical page. Per a PBX manufacturer's configuration example: the port must stay exactly 5060 with UDP (any port translation on NAT breaks the link) or TCP is used, on quality problems a single codec is left — G.711 a-law, and tone dialling follows RFC 2833. A user name is created anyway — it is needed to show your own number and to hide the number. ⚠️ The product is traditionally sold together with Vodafone's internet access: whether the trunk can be ordered on a third-party internet connection is not stated publicly — make that your first question.

The others#

• 1&1 Versatel — the "1&1 Voice (Business) SIP Trunk", registration mode, UDP/TCP or TLS transport, G.722/G.711/G.729 codecs; the operator always expects the dialled number in the +E.164 format. ⚠️ The main blocker: the operator states plainly that the trunk is tied to its own internet access and cannot be connected centrally in a data centre — the product is not suitable for a PBX in the cloud. It does not support CLIP no screening either, so the "Show as" rule will not work.
• NFON — "Nconnect Voice / SIP Trunk Flexx", a trunk specifically "for a third-party PBX", including an uncertified one; indirect evidence points to registration mode. The parameters are handed out in a document ("Merkblatt") through the documentation portal and cannot be seen before a contract: request it from the partner together with the SIP header examples.
• inopla / comdesk — a virtual PBX where the trunk is created as an "endpoint" in the control panel (virtuTEL → Endpunkte): the login, the password and the domain of your particular trunk are shown right there, that is, the SIP host is different for every account. Be sure to enable the "DDI durchreichen" option in the number routing, otherwise only the head number will reach the PBX.
• Outbox AG (foncloud) — a Cologne operator with a cloud PBX and a SIP trunk; the parameters are issued in the control panel. ⚠️ Passing the number in the P-Preferred-Identity header is a hard condition for establishing a call there, and our PBX does not send that header: before ordering, clarify whether outgoing calls will go through without it.
• toplink — "SIP Business". The operator publicly presents encryption of all calls as the standard mode, and no unencrypted variant is visible on its pages. Our PBX does not support TLS for trunks, so ask directly before ordering whether a plain UDP/TCP trunk exists.

International providers with German numbers#

German numbers are sold by DIDWW, Telnyx, Twilio, Plivo, Vonage, Bandwidth, Sinch, Infobip, Zadarma, didlogic and others. This is the default path if the company is registered outside Germany: local operators sign the contract with a German entity.

• The tie of a landline number to its area does not go away. A German landline number is issued with a tie to the calling area where the subscriber is located, and the operator is obliged to check it — at international providers the same thing is arranged as "proof of a local address or business registration". Buying a Cologne number while sitting in another country will not work with a reputable provider.
• Only your own number may be shown on outgoing calls. German law allows presenting only a number you hold the right to use, and only from the German numbering range; the exception for foreign numbers applies solely to businesses, with proven rights to the number, for calls to the country of that number and with prior notice to the operator.
• Emergency calls are a separate service at international providers, and more often are not available at all: calling 112/110 from Germany requires a local operator or an explicitly ordered Emergency service.
• The modes differ. Telnyx and DIDWW offer both registration and a static IP endpoint; Twilio Elastic SIP Trunking works by IP only — the static address of your PBX will be needed; Zadarma uses registration. The cards for Zadarma, Telnyx and Twilio are already in the catalogue when adding a provider: the server address and the mode are filled in for you.

Number porting#

• The right to port is set by law (§ 59 TKG). The transferring and the receiving operators are obliged to cooperate and not to delay the port; the interruption of service may not last longer than one working day, and for every further working day of downtime the subscriber may claim €10 or 20 % of the monthly fee (whichever is greater). While the port is running, the fee at the previous operator is halved after the contract ends.
• The application is filed with the new operator — the one you are moving to: a signed "Portierungsformular" plus the termination of the contract with the previous operator. reventix warns separately that the customer details at the old and the new operator must match character by character, and advises scheduling the porting date on a Monday to Thursday.
• The head number and the Durchwahl block are ported — including when moving from an old ISDN connection to a SIP trunk. The digits the subscriber appended himself to the issued number, however, create no rights and are not ported.
• When the venue relocates, the landline number does not stay with you automatically: it is tied to the calling area, not to the subscriber. Changing the address within the area is a transfer of the service; changing the area usually means a new number. Keep the address at the operator up to date after a move as well: the location for emergency services is determined by exactly that address, and a PBX in the cloud does not change it.

Rules and documents#

If it does not work#

1. "No registration at the provider" while the login and password look correct. Almost certainly the control-panel password was entered: at easybell, peoplefone, dus.net and reventix the SIP password is separate. The second typical case is the form of the login: at reventix it is padded with three zeros on the left, at dus.net the User ID comes with leading zeros.
2. The server address is mixed up. At easybell voip.easybell.de is the trunk while pbx.easybell.de is their cloud PBX; at fonial the main address is sip.plusnet.de while proxy01.sip.plusnet.de is the outbound proxy; at reventix, next to reg.sipbase.de, third-party instructions mention the address for the encrypted connection. Half of all "it does not register" cases come from here.
3. Telekom: silence without errors. With the DeutschlandLAN SIP-Trunk the signalling works over TCP only — the "Transport" field must be set to TCP. The current CompanyFlex on a third-party access requires TLS and SRTP, which we do not have: that product cannot be connected.
4. Vodafone: calls do not go through although the address is correct. The mode is by IP: the operator must see the static address of your PBX (the "Server" tab), and with UDP the port must stay exactly 5060.
5. peoplefone: incoming calls arrive but are not sorted out by number. The line has not been switched to the "SIP-TRUNK" type — ask support to do it.
6. The wrong number is shown on outgoing calls. sipgate, Placetel and Outbox read the number from the P-Preferred-Identity / P-Asserted-Identity headers, which our PBX does not send. Placetel will substitute an arbitrary number from your account. The workaround is to make the number you need the trunk's default number at the operator.
7. Incoming calls arrive but no line is selected. At fonial and in Placetel's example the trunk login arrives instead of the number: the call is accepted as "from this provider", but numbers cannot be told apart by lines — "The provider sends its login instead of the number".
8. Calls to two numbers land on the same line. The last seven digits of numbers from different calling areas coincided. Separate such lines manually, or take numbers from a single area plus a Durchwahl block.
9. The call connects but there is silence — the codecs did not negotiate: our PBX speaks G.711 and G.722, while dus.net and inopla do not describe their codec set publicly (ask support to enable G.711). The second simultaneous call drops — the tariff channels ran out: at easybell that is 2, 10 or 50 calls, at the others it is set by the contract.

Causes that do not depend on the country are in "If calls do not go through"; the SIP trace in the trunk menu is described there as well. Checking the address and the port is "Test connection".

FAQ#

Can I get a German number if the company is registered outside Germany?#

At a local operator — hardly: the contract is signed with a German entity, and a landline number is issued with a tie to the calling area where the venue is located, a tie the operator is obliged to verify. None of the operators we checked describes a public procedure for foreign companies. The practical path is an international provider with +49 numbers; it will ask for documents on a landline number too.

Which operator should I choose if nothing is connected yet?#

easybell. It publishes a ready Asterisk configuration with our codecs, its tariff starts at €1.95 per month with no commitment, it has number blocks for all areas and — uniquely in the country — a switch for the incoming number format in the control panel: setting it to "E.164" removes the main cause of routing problems. The fallbacks are sipgate trunking and Placetel.

Why is Deutsche Telekom not suitable?#

The current CompanyFlex product requires mandatory encryption of signalling and calls on a third-party access, and our PBX does not support TLS and SRTP for trunks. The older DeutschlandLAN SIP-Trunk is technically connectable (the TCP transport is required) but is closed to new sales. If you already have such a trunk, connect it — starting by switching the transport.

Can I show a different number on outgoing calls?#

Only another number of your own, one you hold the right to use, and only from the German numbering range — that is what the law requires. There is a second, technical limitation too: sipgate, Placetel and Outbox read the number from service headers that our PBX does not send, so with them the call will go out with the trunk's default number.

How many channels does a venue need?#

A channel is one simultaneous call. A restaurant usually needs 2–3: one is busy with a call to a guest, the second with a courier or a supplier. At easybell the channels are firmly tied to the tariff (2/10/50), while contract-only operators negotiate the number of channels separately — ask for it before signing.

Sources#

All links checked on 09.09.2026.

Operators

• easybell — configuring Asterisk (a complete configuration example): https://www.easybell.de/hilfe/telefon-konfiguration/ip-telefonanlagen-fuer-unsere-sip-trunks/antwort/asterisk-telefonanlagen/
• easybell — in which format incoming calls are signalled: https://www.easybell.de/hilfe/telefon-konfiguration/allgemein/rufnummernformat-bei-eingehenden-anrufen/
• easybell — SIP Trunks: tariffs and channels: https://www.easybell.de/business/sip-trunks/
• sipgate — configuring Asterisk for sipgate trunking: https://help.sipgate.de/trunking/en/wie-konfiguriere-ich-asterisk-fuer-sipgate-trunking
• Placetel — connection via PJSIP: https://www.placetel.de/hilfe/sip-trunking/anbindung-mit-asterisk-pjsip
• Placetel — PBX connection checklist: https://www.placetel.de/hilfe/sip-trunking/anbindung
• dus.net — SIP information & settings: https://www.dus.net/en/sip
• fonial — fonial Trunking mit Asterisk: https://www.fonial.de/hilfe/trunking/konfiguration-der-telefonanlagen/asterisk
• peoplefone — SIP TRUNK: https://support.peoplefone.com/deu/peoplefone-sip-trunk/
• reventix — wiki: connecting equipment to a SIP trunk: https://wiki.reventix.de/konfigurationshilfen:avm:fritzbox
• reventix — SIP-Trunk for a telephone system: https://www.reventix.de/produkte/sip-trunk
• Deutsche Telekom — DeutschlandLAN SIP-Trunk, technical documentation: https://geschaeftskunden.telekom.de/internet-dsl/tarife/companyflex/sip-trunk/sip-trunk-technische-unterlage
• Telekom CompanyFlex — use on another provider's access: https://hilfe.companyflex.de/de/einrichtung/anschalteszenarien/nutzung-eines-ip-anschlusses-eines-anderen-anbieters-am-companyflex
• 1&1 Versatel — setting up 1&1 Voice SIP: https://www.1und1.net/hilfe/sip
• NFON — Nconnect Voice / SIP Trunk Flexx: https://www.nfon.com/de/produkte/enablement/nconnect-voice
• inopla / comdesk — SIP-Trunk: https://www.inopla.de/blog/sip-trunk/
• Outbox AG — technical notes on configuring a foncloud SIP trunk: https://faq.outbox.de/geschaeftskunden/foncloud/technische-fragen/technische-hinweise-zur-konfiguration-eines-foncloud-sip-trunks/
• toplink — SIP Trunking: https://www.toplink.de/loesungen/sip-trunk/
• Vodafone — rules for presenting foreign numbers for business customers: https://www.vodafone.de/business/hilfe-support/hinweis-fuer-das-aufsetzen-auslaendischer-rufnummern.html

Regulator and law

• Bundesnetzagentur — allocation of landline numbers and the tie to the calling area: https://www.bundesnetzagentur.de/DE/Fachthemen/Telekommunikation/Nummerierung/ONRufnr/Zuteilungsverfahren/artikel.html
• Bundesnetzagentur — the length of landline numbers: https://www.bundesnetzagentur.de/DE/Fachthemen/Telekommunikation/Nummerierung/ONRufnr/Laenge/L%C3%A4nge_ONRufnr_Basepage.html
• Bundesnetzagentur — better protection against number manipulation from 01.12.2022: https://www.bundesnetzagentur.de/SharedDocs/Pressemitteilungen/DE/2022/20221129_TK-Aktueller-Hinweis.html
• § 5 TKG — notification duty: https://www.gesetze-im-internet.de/tkg_2021/__5.html
• § 59 TKG — operator switching and number porting: https://www.gesetze-im-internet.de/tkg_2021/__59.html
• § 120 TKG — transmission of the calling number: https://dejure.org/gesetze/TKG/120.html
• § 164 TKG — emergency calls: https://www.gesetze-im-internet.de/tkg_2021/__164.html

Related pages#

• Connecting a provider's number to the PBX — the general steps, trunk modes, network, testing the connection.
• Setting up SIP phones — how to connect desk phones and DECT handsets to extensions.