An outgoing call is a call your team starts to another number. With business SIP telephony, a softphone, IP phone or PBX sends that call through your provider to the destination network. An incoming DID number gives customers a way to call you; it does not, by itself, configure your outbound route.
This guide covers the practical setup: account access, dialling rules, Caller ID and a first test call. Start with KataTelecom IP telephony for a SIP account, or SIP Trunking when connecting an existing PBX.
How an outgoing SIP call reaches the recipient
- Your device or PBX: selects the outgoing account and applies its dialling rule to the number entered.
- The SIP provider: checks account access and sends an eligible call through the configured external route.
- The destination network: delivers the call to the recipient’s mobile, landline or other supported destination.
Call setup and conversation audio are separate parts of the connection. A phone can show a connected call while an audio-path problem prevents one side from hearing the other. Test both the call result and sound in each direction.
What to confirm before configuring the phone
- Account settings: SIP server, authentication method, username and any required proxy or transport settings. Use the values issued for your account.
- Destinations: the countries, number prefixes and fixed or mobile networks your team needs to call.
- Caller ID: the business number you want recipients to see and the verification or route requirements for presenting it.
- Usage and capacity: the selected rate, payment conditions and expected simultaneous calls. Call volume and concurrent-call capacity are different requirements.
- Device: a compatible softphone, SIP phone or PBX, plus a working microphone, speaker or headset.
Keep credentials private. Your personal account and the team provide the exact settings and conditions for the selected offer; a generic configuration example is not a substitute for those values.
Set up and test your first outgoing call
- Configure one account first. Add the supplied SIP settings to one phone or PBX trunk. Check registration if the selected configuration requires it; some trunks use IP-based authentication instead.
- Select the outbound route. On a PBX, confirm that the extension is allowed to use the intended trunk. A registered extension alone does not prove that external calling is configured.
- Check the dialled number. Start with a complete international number and confirm the exact format your route accepts. A PBX access digit, a domestic prefix and the destination country code serve different purposes; inspect what the PBX actually sends.
- Confirm Caller ID. Use the number approved for that account and route. Check the result on a phone your team controls.
- Check the conversation. Make an agreed test call. Listen in both directions and verify that ending the call releases it normally.
- Review the call record. Match the destination, time, duration and result with the account’s call history. Then test the other destination types your business uses before moving regular traffic.
Change one setting at a time and keep a note of the test result. For mobile app behaviour and network changes, see the Mobile VoIP guide.
Caller ID and the number used for callbacks
Decide which business number recipients should see and where a return call should go. Confirm that you can use that number as outgoing Caller ID on the selected route. Buying an incoming number and entering it into a phone’s display field are separate from approval of its outbound presentation.
The receiving network can also affect how the call is shown. Test the intended destinations rather than assuming every handset will display the same number or business name. Our Caller ID settings page explains the service; ask the team to confirm the requirements for your chosen number.
If you also need customers to call back, choose an appropriate virtual phone number and set up its incoming route separately. Forwarding that incoming call to another phone is a different call flow, covered in the call forwarding guide.
When outgoing calls do not work
First identify the stage that failed: access to the SIP service, call setup, delivery to one destination, or conversation audio. A response code is a clue, not a complete diagnosis.
- No response from the service: check the issued server address, connectivity and configured transport. Confirm that the request reaches the expected service before changing the dialling rule.
- Repeated 401 or 407: these are authentication challenges. A challenge followed by a successful authenticated request can be normal; a repeated failure needs a check of the account’s authentication settings.
- 403: the request was refused. Check the route and account conditions with support instead of repeatedly retrying unchanged settings.
- 404: check the destination and the domain or number format sent by the phone or PBX.
- 488: check the offered media settings and supported codecs with the provider.
- Connected call, missing or one-way audio: check microphone permissions, the selected audio device and the RTP/media path through the firewall or NAT.
The code meanings are documented in RFC 3261. For Asterisk installations, its PJSIP troubleshooting guide and NAT configuration guidance help separate signalling and media issues. Follow the configuration for your own equipment.
When asking for help, provide the test time with time zone, destination, device or PBX, call result and any available call identifier. Share logs through the team’s support channel after removing passwords and authentication data.
Choose the offer for your business
IP telephony connects your team over the internet. It allows you to make unlimited outgoing calls, which saves the company’s budget.
You are free to choose the rate yourself based on your business needs. Our guide to comparing the cost of international business calls explains how destinations, billed minutes and the complete route affect that choice.
Confirm the selected offer’s billing rules, any package limits and concurrent-call capacity in your personal account before moving traffic. The general service description does not specify those account-level conditions. Use the current outgoing call rates for destination pricing.
You can choose a compatible mobile or desktop softphone, IP phone or PBX for your team. We will recommend the most suitable rate for you, and provide you with 24-hour technical support.
Outgoing calls: common questions
What is the difference between outgoing and incoming calls?
Your team starts an outgoing call to another number. An incoming call starts with someone calling your public contact number. Configure and test the two directions separately.
Do I need a virtual number to make outgoing calls?
Number and Caller ID requirements depend on the destination, account and route. Confirm the required setup with the team. An incoming DID does not automatically enable outbound calls or approve its number as outgoing Caller ID.
Can I use my existing PBX?
A compatible PBX can send outgoing calls over SIP. Confirm the account settings, extension permissions, outbound dialling rule and capacity, then test the destinations you need.
Why can I receive calls but not make them?
The two directions use separate routing and permissions. Check the outbound account, the PBX’s trunk selection, the number it sends and the selected offer’s conditions.
Why is there no sound even though the call connects?
Call setup can succeed while audio fails. Check the device’s microphone and speaker selection, permissions and the media path through NAT or a firewall. Give support a specific test time and call result if the issue continues.
Where can I check the price of an outgoing call?
Use the current destination rates and confirm the selected route’s billing rules in your personal account. Number rental, any setup charges and other call legs should be checked separately.
Ready to configure your route? Contact KataTelecom with your PBX or device, intended destinations, Caller ID and expected simultaneous calls.