A SIP trunk in Saudi Arabia is a digital phone line that a carrier (stc, Mobily or Zain) delivers over an IP connection and that plugs straight into an on-site IP PBX, replacing analog landlines or E1 circuits. From the carrier you request the numbers, the channel count and the technical connection details. On the PBX you configure the trunk, inbound and outbound call routes and the network settings. The 9200 unified number is a separate service whose calls are routed to your lines in whatever way the carrier allows.
Key takeaways
- stc and Zain both list a business SIP trunk service on their websites, and Cisco has published an application note on connecting a PBX to a Mobily SIP trunk.
- stc states its service scales from 5 to 30,000 channels over fiber, point-to-point microwave or fixed wireless access (4G/5G).
- Zain states that 9200 unified numbers, 800 toll-free numbers and geographic numbers are available with its SIP trunk.
- Connection settings differ by carrier: authentication method, voice codec, number format and fax handling.
- The channel count should never exceed the concurrent calls your PBX can carry.
In this article
- What is a SIP trunk and how does it connect to the PBX?
- SIP trunk in Saudi Arabia: what each carrier publishes
- The 9200 unified number and your PBX
- What to request from your SIP trunk provider
- What the PBX needs for a SIP trunk connection
- Example: what stc’s SIP configuration document specifies
- Common mistakes when connecting a PBX to a SIP trunk
- How AZEIDK helps with SIP trunk connection
- Frequently asked questions
- Conclusion
What is a SIP trunk and how does it connect to the PBX?
SIP (Session Initiation Protocol) is the protocol most modern PBX systems use to set up calls over a network. A SIP trunk means the carrier hands over your phone lines in digital form on a data link, so calls go in and out of the PBX without analog ports or E1 cards. How SIP trunks compare with analog and E1 lines when choosing a PBX is covered in our IP PBX for businesses guide, and moving off an old system in analog PBX to IP migration.
The practical difference from traditional lines is that call capacity is no longer tied to physical wires. You order a number of channels (each channel carries one call at a time) and a block of direct numbers (DIDs), each of which can be routed to an extension or a department.
The PBX systems AZEIDK supplies as part of its network and communication systems, such as Yeastar, Grandstream and Avaya, all work with SIP trunks. Yeastar’s administrator guide, for example, describes three types: a register trunk using a username and password, a peer trunk authenticated by IP address and port, and an account trunk for connecting other devices.
SIP trunk in Saudi Arabia: what each carrier publishes
The table summarises what we found in official sources. It includes no prices; contract terms come from the carrier directly.
| Carrier | What is published | Source |
|---|---|---|
| stc | Enterprise voice over VoIP, from 5 to 30,000 channels, with SIP backup, multi-branch connection to a central system, and integration with platforms such as Microsoft Teams. Access over fiber, microwave or 4G/5G. | stc SIP Trunking page |
| Zain | IP voice based on the SIP standard, with geographic numbers (011, 012, 013 and others), 9200 unified numbers and 800 numbers, direct inward and outward dialing (DID/DOD) and multiple channel options. | Zain KSA SIP Trunk page |
| Mobily | An application note published by Cisco (April 2020) on connecting a Cisco PBX to a Mobily SIP trunk through Cisco UBE. | Cisco, Mobily SIP Trunk application note |
The 9200 unified number and your PBX
The 9200 unified number is a special service number that customers can dial from anywhere in the Kingdom, and from abroad by adding Saudi Arabia’s country code. On its service page, stc says it lets you distribute, forward and time-route calls, handle queued calls and busy lines, cap the number of calls and record them. stc takes requests through 909 or any of its branches.
The key question for whoever configures the PBX is where the 9200 number terminates. Does it land on DIDs on the same SIP trunk, or on other lines? The stc page does not say, while Zain lists 9200 as part of its SIP trunk offer. Ask the carrier in writing which number or numbers 9200 calls will arrive on, so you can program a clear inbound route for them, such as an auto attendant or a customer service group.
What to request from your SIP trunk provider
Before you sign the order, collect these details from the carrier. They are what the PBX engineer needs to finish the connection:
- Numbers: the main number, the DID block, any 9200 or 800 number, and where those calls terminate.
- Channel count: based on the highest expected number of simultaneous external calls.
- Link type: fiber or wireless, whether the link is dedicated to voice or shared with internet, and the customer-side IP address.
- Authentication: registration with a username and password, or IP-based.
- Server, port and transport: the SIP server address or domain, the port, and UDP or TCP.
- Voice codec: for example G.711 A-law or G.729, and the priority order.
- DTMF method: a mismatch here breaks the auto attendant.
- Number format: whether the PBX must send numbers in international +966 format or local format.
- Fax: G.711 pass-through or T.38.
- Backup: what happens to calls if the link goes down.
What the PBX needs for a SIP trunk connection
Trunk definition
The trunk is added on the PBX using the type the carrier specifies (register or IP), with the server, port and codec details.
Inbound and outbound routes
Each DID is routed to its destination: an extension, a ring group or an auto attendant. Outbound rules are set so numbers go out in the format the carrier expects.
Network and firewall
Some carriers require a static route to their server network and specific ports opened for voice. Some projects place a session border controller (SBC) between the PBX and the carrier, as in Cisco’s Mobily application note, which uses Cisco UBE.
Capacity
The channel count must match what the PBX can handle. For example, Yeastar lists the P550 at 25 concurrent calls, so 30 channels on it would add nothing. The switches powering the phones also need a correct power calculation, as explained in our article on PoE switch power budget.
Example: what stc’s SIP configuration document specifies
stc has published a SIP configuration document on its website for connecting a customer PBX. The settings below are taken from it as an example of the data you receive. They may differ by service and contract, so confirm them with the carrier before programming:
| Item | Value in the stc document |
|---|---|
| Transport and port | UDP, port 5060 |
| Voice codec | G.711 A-law first (ptime 20), then G.711 U-law |
| Trunk keep-alive | stc sends OPTIONS and the PBX replies 200 OK; the PBX does not send its own OPTIONS |
| Number format | International format: +966, then area code, then number |
| Fax | G.711A pass-through |
| Network | Static route to the SIP server subnet via the stc gateway |
In Cisco’s Mobily note (2020), the PBX registered to Mobily’s server over UDP, with G.711 A-law and G.729 codecs and DTMF per RFC 2833. These are examples from published documents, not a substitute for the carrier’s data for your contract.
Common mistakes when connecting a PBX to a SIP trunk
- One-way audio caused by NAT or firewall settings.
- An auto attendant that ignores keypresses because the DTMF method does not match.
- Outbound calls rejected because numbers are not sent in the required format.
- 9200 calls landing on a single extension because their terminating number has no route.
- Ordering more channels than the PBX can carry, or fewer than the real need.
How AZEIDK helps with SIP trunk connection
AZEIDK supplies IP PBX systems in Saudi Arabia under IP telephone systems. Work starts with technical selection of systems and equipment based on channel and extension count, followed by programming and systems integration, including the SIP trunk and call routes, using the data the carrier provides. The contract, numbers and channels themselves are ordered from the carrier directly.
Frequently asked questions
Can I connect my PBX to an stc SIP trunk?
Yes. stc offers SIP Trunking for businesses and has published a configuration document describing the PBX-side settings.
Does a 9200 unified number work with a SIP trunk?
Zain lists 9200 as available with its SIP trunk. With any carrier, ask which numbers 9200 calls terminate on so you can program a route for them on the PBX.
How many SIP channels do I need?
Enough for your peak simultaneous external calls, without exceeding the concurrent call capacity of your PBX.
What is the difference between a register trunk and an IP trunk?
With a register trunk the PBX registers to the carrier with a username and password. With an IP (peer) trunk the carrier trusts the PBX’s IP address without registration. The carrier decides which type applies.
Do I need an SBC?
Not always. It depends on the carrier’s requirements, the network design and the site’s information security policy.
Does fax work over a SIP trunk?
It works when the method matches the carrier’s. The stc document, for example, calls for G.711A fax pass-through.
Conclusion
A successful SIP trunk connection in Saudi Arabia comes down to a clear data sheet from the carrier and a PBX configured to match it. Get the numbers, channels, authentication, codec and number format in writing, settle the 9200 route before programming, and make sure the PBX capacity fits the channel count.
Getting ready to connect your PBX to a SIP trunk? Send us the carrier’s details and your extension count, and our team will review PBX and network readiness. Talk to the AZEIDK team
