Channels
Voice: connecting your telephony over SIP
How your existing phone lines connect to Telonic, what stays with your carrier, and what your telephony team provides.
On this page
- What stays the same for you and your callers
- How a call reaches the agent
- The technical detail
- Choosing which calls go to the agent
- Outbound calls through your carrier
- Handing a call to your team
- What your telephony team provides
- Systems this works with
- How a line goes live
- In practice
- What your team controls
- Related
Your customers keep calling the numbers they already know. Telonic connects to your phone system over SIP (the standard protocol business phone systems use to set up and connect calls), through the carrier you already use. Your numbers, your carrier contract and your call routing stay yours, and nothing changes for the people calling you. This page explains how the connection works, what your telephony team provides, and how a line goes live.
What stays the same for you and your callers
- Your published numbers do not change. Customers dial the same numbers they use today.
- Your carrier stays your carrier. The contract, the telecoms licence and the numbers remain with your carrier. Telonic does not sell numbers or call minutes, and does not act as a carrier.
- Your routing stays in your hands. Your phone system decides which calls reach the agent, and when.
- Your contact centre platform stays in place. Telonic works alongside it rather than replacing it.
Because the routing sits in your phone system, you can send calls back to your team at any time by changing a routing rule on your side.
How a call reaches the agent
When a customer dials your number, the call arrives at your carrier exactly as it does today. Your phone system then applies the routing rules you have set. Calls you have chosen to route to the agent travel over a SIP trunk (a virtual phone line that connects two phone systems over a network) to the Telonic SIP endpoint (the address in your deployment that receives SIP calls) in your chosen region.
The agent answers, holds the conversation, and looks up what it needs in your systems while the caller is on the line. When the caller needs a person, the agent passes the call back to your phone system, which delivers it to the team or number you specified.
Your customer
Your carrier
Contract, licence and numbers stay here
Your phone system
Your routing rules
10pm to 8am → agent
Queue over threshold → agent
All other calls → your team
Telonic, in your chosen region
- 1Call as today
- 2SIP trunk: signalling over TLS, audio as a separate stream
- 3Transfer to your team, with a written brief
Your customer
Your carrier
Contract, licence and numbers stay here
Your phone system
Your routing rules
10pm to 8am → agent
Queue over threshold → agent
All other calls → your team
Telonic, in your chosen region
The technical detail
A SIP call has two parts. Signalling, carried by SIP, sets the call up, transfers it and ends it. Media, the audio itself, travels separately as an RTP stream (the real-time protocol that carries voice across IP networks). Telonic receives both from your side of the connection, usually from your session border controller (the device or service at the edge of your phone network that manages and secures SIP traffic), your IP phone system (a phone system that carries calls over your data network), or your contact centre platform (the software that queues and routes calls to your team).
Telonic supports standard SIP, which enterprise phone systems use, so no proprietary equipment is needed on your side. The SIP endpoint and the handling of call audio for your deployment run in the region you choose. See Hosting and data residency.
| Item | Telonic's approach |
|---|---|
| Protocol | Standard SIP, received from your carrier trunk, session border controller, phone system or contact centre platform |
| Signalling security | SIP over TLS (Transport Layer Security), version 1.2 or higher |
| Media security | Call audio is encrypted with SRTP (Secure RTP, the encrypted form of the audio stream) by default. Unencrypted audio is used only over a private network connection, at your written request |
| Authentication | IP allow-listing (accepting traffic only from your agreed addresses) and SIP digest credentials (a username and password exchanged securely during call set-up). Mutual TLS, where both sides present certificates, is supported |
| Audio codecs | G.711 A-law and µ-law, G.722 and Opus (codecs are the formats used to compress call audio) |
| Keypad tones | DTMF (the tones a phone keypad sends) as RFC 4733 telephone events by default, with SIP INFO supported |
| Transfer method | SIP REFER (which hands the call back to your phone system) by default, or a bridged transfer where the brief must be delivered by voice first. A bridged transfer uses a second trunk channel for the length of the call |
| Where it runs | In your chosen hosting region, alongside the rest of your deployment |
| If the endpoint cannot be reached | Your session border controller or phone system routes calls to your team instead. Telonic supports SIP OPTIONS health checks (regular messages that confirm the other side is available), so your side detects this automatically |
Choosing which calls go to the agent
You decide which calls the agent takes, and the rules live in your phone system or contact centre platform. Most organisations start with one line or one time window, then extend.
| Routing pattern | How it works | A typical first use |
|---|---|---|
| A dedicated number or line | Every call to that number goes to the agent | An enquiry line for one property launch |
| Out of hours | Your system routes calls to the agent outside your team's working hours | Reservations overnight and on public holidays |
| Overflow | Calls go to the agent when your queue passes a threshold you set | A claims line during a storm or major event |
| A menu option or queue | One option or queue in your existing menu routes to the agent | "Check the status of a claim" |
| Outbound | The agent places calls through your trunk, showing your number | Appointment reminders and follow-ups |
Outbound calls through your carrier
Outbound calls placed by the agent leave through your own trunk and show your own number, so they are carried and billed under your existing carrier arrangement. Before any outbound call, the agent checks the customer's contact preferences and the calling hours you have set, within those the rules allow. See Contact rules for outbound messages and calls.
Where your carrier supports it, your business name can be registered to appear on the recipient's phone. See Keypad input, voicemail detection and caller ID.
Handing a call to your team
When a call needs a person, the agent transfers it to the number, extension or queue you specify for that situation. The person taking over receives a written brief with the call: who the caller is, what they asked, what has been established and what is outstanding. The caller does not have to repeat themselves. The brief is delivered into the system your team works in: your contact centre platform, your CRM or the Telonic console. The call itself carries a conversation reference in a SIP header, so your system can link the call to the brief.
Transfers, scheduled callbacks and overflow are covered in Transfers, callbacks and overflow.
Calls are recorded only once the caller has given consent. The consent request is part of the call flow, and the caller's answer is recorded against the conversation. Before consent, only the consent answer is recorded; nothing else from that part of the call is stored. See Call recording and consent.
What your telephony team provides
| What we need | Why | Who usually owns it |
|---|---|---|
| A SIP route from your session border controller, phone system or contact centre platform to the Telonic SIP endpoint | This is the connection calls travel over | Telephony or network team |
| The numbers to route, and the routing rules | Decides which calls reach the agent, and when | Contact centre operations |
| Transfer destinations for each situation | Decides where a call goes when it needs a person | Contact centre operations |
| Network access for signalling and audio between your side and the Telonic endpoint | Lets the two systems reach each other securely. Telonic provides fixed IP addresses, port ranges and a domain name for your allow-list | Network or security team |
| The number of simultaneous calls the trunk should carry | Sets how many calls can reach the agent at once | Telephony team, with your carrier |
| Test numbers and a test window | Lets both teams test calls before customers reach the agent | Telephony team |
Telonic provides the SIP endpoint in your chosen region, the connection details, a configuration guide for your phone system, and an engineer who tests the connection with your team. The agent handles many calls at the same time. The number of simultaneous calls for your deployment is agreed during implementation, alongside the capacity of your trunk. A bridged transfer uses a second trunk channel for the length of the call, so allow for this when sizing the trunk.
Systems this works with
Telonic connects over SIP to the carriers and phone systems enterprises in the region use, including e&, stc, Vonage, Cisco and Zoom.
Where you run a contact centre platform, such as Genesys, NICE, Five9 or Amazon Connect, the agent works alongside your existing platform and routing. Your platform keeps deciding where each call goes.
If your carrier or phone system is not listed, it will usually still connect, because the connection uses standard SIP. Your telephony team and ours confirm this during the first technical call.
How a line goes live
- Map. Together, we map your current call flows: which numbers, which menu options, which queues, and where calls go today.
- Agree. You choose which calls the agent takes first, and where it hands calls back to your team.
- Connect. Your telephony team configures the SIP route and network access. We configure the endpoint for your deployment.
- Test. Both teams test inbound calls, outbound calls, transfers, keypad input and the recording consent step, using test numbers, before any customer reaches the agent.
- Go live. Your team switches the routing rule for the first line or time window.
- Extend. Adding another line, queue or time window is a routing change on your side, because the connection is already in place.
In practice
A hotel group in Riyadh routes its reservations line to a Telonic agent overnight.
- The group's numbers are on stc lines, handled by a Cisco phone system. Neither changes.
- The telephony team adds one rule: between 10pm and 8am, calls to the reservations number route over the SIP trunk to the agent.
- At 1:40am, Faisal calls to add a night to his stay. The agent finds his reservation, confirms the room is available at his booked rate, and extends the stay.
- At 2:15am, a guest calls about a lost passport. The agent recognises that this needs a person, and transfers the call to the night manager's number with a brief.
- At 8am, the routing rule ends. Calls go to the reservations team as usual, and every overnight conversation is already in the customer record.
What your team controls
- Which numbers, menu options and queues route to the agent.
- The hours and overflow thresholds that trigger routing.
- The transfer destination for each situation.
- The number outbound calls show.
- Switching routing back to your team, at any time, in your own phone system.
Related
- Voice: inbound and outbound callsChannels
- Voice: transfers, callbacks and overflowChannels
- Telephony and carriersIntegrations
- Contact centre platformsIntegrations
- What we need from your IT teamIntegrations
Product names and logos are trademarks of their owners. Their mention shows systems Telonic connects to and does not imply partnership or endorsement.