--- title: Connect Avaya IP Office to a Kenyan SIP Trunk (Sautikit) description: >- Add a Sautikit Kenyan number to Avaya IP Office as a SIP Line: ITSP proxy, send port 5080, SIP credentials, SIP URI, incoming call route and ARS. summary: >- In IP Office Manager, create a SIP Line to sip.sautikit.com with send port 5080 over UDP, add your Sautikit SIP credentials, a SIP URI, an incoming call route and ARS. date: 2026-11-05T00:00:00.000Z type: guide --- ## Summary An Avaya IP Office system can use a Kenyan Sautikit number as its SIP trunk. In IP Office Manager, create a **New SIP Line** with ITSP Domain Name and ITSP Proxy Address `sip.sautikit.com`, Layer 4 Protocol UDP and **Send Port 5080**. On **SIP Credentials**, add your Sautikit SIP username (as User name and Authentication Name) and password with **Registration required** ticked. Add a **SIP URI** that uses that registration, an **Incoming Call Route** for the line group, and a short code or ARS entry for outbound calls in `+254` format. You also need enough **SIP Trunk Channels** licences. The number costs KES 500 a month, inbound calls are free and outbound calls cost KES 3.00 a minute billed per second. ## What you need - **A Sautikit number** (KES 500 a month, live in seconds) and a SIP credential for the IP Office: in the [Sautikit console](https://app.sautikit.com), open the number, go to **SIP & Devices**, click **+ Add device**, label it "IP Office", and keep the username, password and the device's **extension**. - **IP Office Manager** with admin rights (R11 or R12). - **SIP Trunk Channels licences.** Avaya licenses SIP trunk calls by channel; check **License > SIP Trunk Channels** has instances for the simultaneous calls you need. On Server Edition the setting is **SIP Trunk Sessions**. ## Sautikit settings at a glance | IP Office field | Value | |---|---| | ITSP Domain Name | `sip.sautikit.com` | | ITSP Proxy Address | `sip.sautikit.com` | | Layer 4 Protocol | UDP | | Send Port | `5080` | | Listen Port | leave at 5060 | | User name | your Sautikit SIP username | | Authentication Name | your Sautikit SIP username (same value) | | Password | your Sautikit SIP password | | Registration required | ticked | | Codecs | G.711 U-law, G.711 A-law first | ## Step 1: check SIP trunks are enabled Under **System > LAN1 > VoIP**, make sure SIP trunks are enabled. If the IP Office sits behind NAT, set the public IP and firewall type under **LAN1 > Network Topology**. ## Step 2: create the SIP Line In the navigation pane, right-click **Line** and choose **New > SIP Line**. Note the line number; you will use it as the incoming and outgoing group below. - **SIP Line tab:** ITSP Domain Name = `sip.sautikit.com`. - **Transport tab:** ITSP Proxy Address = `sip.sautikit.com`, Layer 4 Protocol = UDP, **Send Port = 5080**, Use Network Topology Info = LAN1. Leave Calls Route via Registrar on. - **VoIP tab:** in Codec Selection, put G.711 U-law and G.711 A-law first. Port 5080 is the setting most often missed: IP Office defaults the send port to 5060, and Sautikit listens on 5080. ## Step 3: add the SIP credentials On the **SIP Credentials** tab, click **Add** and enter the User name, Authentication Name and Password from the table. Tick **Registration required**. Set Expiry to 60 minutes or less. ## Step 4: add a SIP URI On the **SIP URI** tab (under Call Details on R12), click **Add**: - **Local URI, Contact, Display Name:** Use Credentials User Name. - **Registration:** the credential from step 3. - **Incoming Group** and **Outgoing Group:** the line number from step 2. - **Max Calls per Channel:** the number of calls you expect at once. ## Step 5: inbound calls **In Sautikit:** open the number and set **Forward** to the IP Office device's extension. Calls to your number now go to the SIP Line. **In IP Office:** create an **Incoming Call Route** with Line Group ID = the line number, and set the destination (a hunt group, auto attendant or user) on the **Destinations** tab. ## Step 6: outbound calls Add a short code or ARS entry that sends external calls to the line group and presents them in E.164: for example, turn `0XXXXXXXXX` into `+254XXXXXXXXX`. Sautikit shows your Sautikit number as caller ID on every outgoing call. ## Step 7: test Merge the configuration, check the line shows as registered in System Status, then call your mobile from an extension and call the Sautikit number from your mobile. ## Avaya Aura and Communication Manager Avaya Aura systems connect to SIP providers through an Avaya SBC for Enterprise (SBCE) rather than registering from Communication Manager itself. On the SBCE, add a **Trunk Server** profile for `sip.sautikit.com`, port 5080, UDP, enable authentication with your Sautikit SIP username and password, and turn on registration. Your Avaya partner will route calls from Session Manager to that SBCE profile. ## Troubleshooting **The line does not register.** Check Send Port 5080, UDP and the proxy address spelling. Check the credential has Registration required ticked. **Registration fails with an authentication error.** User name and Authentication Name must both be the generated Sautikit username, not your phone number. **Calls stop at two or three at once.** Check your SIP Trunk Channels licence instances. **Outbound calls are refused.** Destinations must reach Sautikit in `+254` format, and your wallet needs credit. ## FAQ ### Can Avaya IP Office use a register-based SIP trunk? Yes. The SIP Line's SIP Credentials tab has a Registration required option, and the SIP URI is tied to that registration. ### Which port does Sautikit use? 5080 over UDP at `sip.sautikit.com`. Set it as the SIP Line's Send Port. ### Does Avaya charge for SIP trunks? Avaya licenses SIP trunk calls by channel through the SIP Trunk Channels licence. On Sautikit there is no trunk fee: the number is KES 500 a month, inbound calls are free and outbound calls are KES 3.00 a minute billed per second.