How to Connect Your SIP Trunk (BYOC Setup)

Quick answer

To connect your own SIP trunk in DialerBee, go to Admin console → Gateways → Add Gateway. The form has six tabs. On General, name the gateway, pick the deployment pattern and the transport protocol (UDP by default) and enter the carrier host and port (5060 by default). On Authentication, choose the authentication mode (None by default, with IP-based, Digest Auth and TLS Certificate as the alternatives). On Codecs, keep or change the default order of PCMA, PCMU and G729 and set the DTMF mode, which defaults to RFC 2833. On Media, set SRTP, NAT traversal and the RTP port range. On Routing, set Max Concurrent Calls (100 by default), the CPS limit (20 by default), priority, weight and a failover gateway, then click Test Connection. On Caller ID, set how the identity is presented. Save, then prove it with a real call.

Bring your own carrier means the platform never sits between you and your carrier bill. The BYOC explainer covers why that matters commercially, and the carrier-neutral SIP trunk setup walkthrough covers the parameters to collect from any provider. This guide is the third piece: the sequence of screens and fields in the DialerBee admin console, with the values the form ships with.

Before you start

Collect from your carrier: the signalling host and port, the authentication method they expect, the channel count and calls-per-second limit, the codecs they accept, the DTMF mode, and the numbers authorised to present as caller ID. If you use IP authentication, they also need the platform addresses to allow.

Step 1: Open the gateway form

Go to Admin console → Gateways, subtitled "Manage SIP gateways, trunks, and routing configuration", with four cards above the list: Total Gateways, Healthy, Active Calls and Total CPS. Click Add Gateway, or Add Your First Gateway on a new tenant.

The modal has six tabs: General, Authentication, Codecs, Media, Routing and Caller ID. Work through them in order.

Step 2: General

Gateway Name, a human readable name. Name it by carrier and region so routing stays readable.
Description, optional and internal.
Deployment Pattern, which defaults to A - Direct Carrier, with B - Tenant PBX and C - 3rd-Party Trunk as the alternatives.
Transport Protocol, which defaults to UDP, with TCP, TLS and WSS (WebSocket Secure) available. Prefer TLS wherever the carrier supports it.
Host / IP Address and Port. Port defaults to 5060.

Step 3: Authentication

Authentication Mode defaults to None, with IP-based, Digest Auth and TLS Certificate as the other choices. Pick the one your carrier asked for.

With IP-based authentication the carrier trusts traffic from the platform addresses, so there is nothing to store. With Digest Auth you complete Auth Username, Auth Password and SIP Domain / Realm, and if the carrier expects a registration you turn on Register with Provider, off by default, and leave Registration Interval (seconds) at 3600 unless they asked otherwise.

For TLS, TLS Version offers TLS 1.2 and TLS 1.3, and Verify TLS Cert is on by default. Leave it on. SIP Profile offers Internal, External and NAT, defaulting to External for a carrier-facing trunk.

Step 4: Codecs

Codec Preferences ships in the order PCMA, PCMU, G729. The picker groups codecs as Narrowband (8 kHz), Wideband (16 kHz) and Video (future), and names them the way a carrier does, for example "G.711 A-law (PCMA)". Agree at least one codec in common with your carrier for each destination you call.

DTMF Mode defaults to RFC 2833 (telephone-event), with Inband (audio), SIP INFO and Auto-detect available. Wrong DTMF handling is invisible until an agent cannot navigate an IVR or a keypress opt-out fails, so test it explicitly.

Step 5: Media

This tab decides whether audio actually flows. The defaults suit a public carrier trunk, so change them only for a reason. Transcoding Enabled, on the Codecs tab, is off by default and should stay off unless two sides share no codec.

Comfort Noise Generation, Echo Cancellation and Packet Loss Concealment: all on.
Jitter Buffer Size, a slider from 20 ms to 200 ms in steps of 10, defaulting to 60 ms.
SRTP Mode, default Disabled, with Optional and Required. Use Required with TLS transport when encrypting media.
NAT Traversal, default None, with STUN, TURN and ICE (STUN + TURN). A STUN/TURN Server field appears once you pick one.
RTP Port Range, default 16384 to 32768. These are the ports the carrier must be able to reach.
Media Bypass / Direct Media, off by default. Leaving it off keeps media on the platform, which recording depends on.
Keepalive Interval (s) 30, Session Timer (s) 1800, and an optional Outbound Proxy.

Step 6: Routing and capacity

Size the trunk against your peak, not your average. A predictive campaign places several calls for every free agent, so concurrency at the top of the hour is the number that matters.

Max Concurrent Calls, default 100.
CPS Limit (calls/second), default 20. Set it at or below the carrier limit, because a dialer bursts hardest at campaign start and shift login.
Priority (1 = highest), default 1, accepting 1 to 10.
Weight (1-100), default 100, range 1 to 100, spreading traffic between gateways of the same priority.
Failover Gateway, empty by default: choose the backup trunk here.
Health Check Interval (s) 30, Max Retries Before Failover 3, INVITE Timeout (s) 60.

Then click Test Connection on the same tab, before any campaign is committed to the trunk.

Step 7: Caller ID

Caller ID Type defaults to Auto-detect, with Remote-Party-ID (RPID), P-Asserted-Identity (PAI) and From Header available. Your carrier will tell you which header they read. Privacy Mode defaults to "None - Caller ID visible", with a restricted mode and a full anonymous mode. Default Caller ID Override sets a fallback number for this gateway.

Each gateway also carries its own DIDs and caller IDs. Expand the gateway row to DIDs & Caller IDs and click Add DID to open the Add DID / Caller ID form, which asks for the DID Number and a Jurisdiction from UAE (TDRA), KSA (CITC), Egypt (NTRA), Jordan (TRC), Global or Other, and gives each number a Verify action. Numbers must also be authorised on the carrier account, because carriers drop calls presenting a number that is not provisioned on the trunk.

Defaults and limits

Field	Tab	Default	Accepted values
Deployment Pattern	General	A - Direct Carrier	A, B - Tenant PBX, C - 3rd-Party Trunk
Transport Protocol	General	UDP	UDP, TCP, TLS, WSS
Authentication Mode	Authentication	None	None, IP-based, Digest Auth, TLS Certificate
Codec Preferences	Codecs	PCMA, PCMU, G729	narrowband and wideband codec list
DTMF Mode	Codecs	RFC 2833 (telephone-event)	RFC 2833, Inband, SIP INFO, Auto-detect
Jitter Buffer Size	Media	60 ms	20 to 200 ms, step 10
SRTP Mode	Media	Disabled	Disabled, Optional, Required
Max Concurrent Calls	Routing	100	as shown on the screen
CPS Limit	Routing	20	as shown on the screen
Priority / Weight	Routing	1 / 100	1 to 10 / 1 to 100
Caller ID Type	Caller ID	Auto-detect	Auto-detect, RPID, PAI, From Header

Verify it works

The gateway list columns are Name, Host / IP, Protocol, Pattern, Status, Calls, CPS, MOS and Actions, and the status values are Not checked, Healthy, Degraded and Down. A metric that has not been measured shows a dash rather than a zero, so a dash is not a failure.

Save the gateway and confirm Status moves from Not checked to Healthy.
Expand the row and read the Quality Metrics panel: ASR, ACD, MOS, Latency, Jitter, Pkt Loss and Active.
Place one outbound test call, confirm it completes and confirm audio is two way. One way audio is almost always the RTP port range or a firewall, not the trunk.
Confirm the number that displays is the number you provisioned.
If you record calls, generate the call recordings report and play one back end to end, before the first dispute rather than after it.
Disable the primary gateway and confirm traffic moves to the failover gateway.

Troubleshooting

Every call fails immediately. Authentication mismatch. Check the mode against the carrier configuration, and that they allowed the platform addresses if you chose IP-based.
The call connects but there is no audio, or audio one way. RTP is not getting through. Confirm the RTP port range is open in both directions and that NAT traversal matches your network.
Calls rejected with a caller ID error. The presented number is not provisioned on the carrier trunk. Add it there, and under DIDs and Caller IDs on the gateway.
A wave of failures at campaign start. You are above the carrier calls-per-second limit. Lower the CPS limit or ask the carrier to raise theirs.
Recordings are silent or missing. Media Bypass / Direct Media is on, so media never reaches the recording leg.

Frequently asked questions

Where do I add a SIP trunk in DialerBee?

Admin console then Gateways, then Add Gateway. The form has six tabs: General, Authentication, Codecs, Media, Routing and Caller ID. A Test Connection button sits on the Routing tab.

What transport and port does the gateway form default to?

Transport Protocol defaults to UDP, with TCP, TLS and WSS available, and Port defaults to 5060. Prefer TLS where your carrier supports it, and pair it with SRTP set to Required if you are encrypting media.

Which authentication mode should I choose?

Whichever your carrier asked for. The field defaults to None and offers IP-based, Digest Auth and TLS Certificate. IP-based needs no stored credential; Digest Auth needs a username, a password and a SIP realm, and a registration if the carrier expects one.

What are the default codecs and DTMF mode?

Codec Preferences ship as PCMA, PCMU and G729 in that order, and DTMF Mode defaults to RFC 2833 with the telephone-event payload. Confirm both against the destinations you actually call, not just against the carrier in general.

How do I size the trunk for predictive dialing?

Set Max Concurrent Calls against your peak concurrency rather than your average, and set the CPS limit at or below the calls-per-second limit your carrier gave you. The form defaults are 100 concurrent calls and 20 calls per second.

How does failover between trunks work?

Choose a Failover Gateway on the Routing tab and leave Health Check Interval, Max Retries Before Failover and INVITE Timeout at values you have agreed with your carrier. Then prove it: disable the primary gateway and confirm calls move to the backup.