SIP Response Code Lookup

A PBX receives an INVITE and immediately acknowledges it while it searches for the destination extension.

The destination phone has been located and is now ringing. The caller's device generates a local ringback tone.

A call to an employee's desk phone is being forwarded to their mobile number because call forwarding is enabled.

A call enters a queue because all agents are on other calls. The caller hears hold music while waiting.

A SIP trunk provider sends early media with an in-network ringback tone or announcement before the far end answers. The SDP body establishes the media path for the early audio.

An INVITE was forked to a user's desk phone and mobile. The mobile answers first, so the PBX sends 199 to terminate the early dialog with the desk phone.

A phone answers an incoming call. The 200 OK includes an SDP body with the answering device's media address and codec selection.

A desk phone successfully registers with the PBX. The Contact header confirms the registered binding and its expiration.

A user initiates a call transfer. The PBX accepts the REFER request and will process the transfer asynchronously.

A phone subscribes to presence updates for a colleague but the server indicates it will not send NOTIFY messages for this subscription.

A user has multiple registered devices. The server returns all of them and lets the caller's UA choose which to try.

A company migrated its SIP trunk to a new provider. The old provider returns 301 with the new SIP URI so that the calling PBX can update its routing.

An employee has call forwarding enabled to their mobile. The PBX redirects the INVITE to the mobile number via the Contact header.

A registrar rejects a direct INVITE and instructs the caller to route through the outbound proxy instead.

A voice call cannot be completed, but the server indicates that a voicemail service is available at an alternative URI.

A phone sends an INVITE with a malformed SDP body where the Content-Length does not match the actual body size. The server rejects it.

A desk phone sends a REGISTER without credentials. The registrar challenges it with a nonce. The phone will re-send with an Authorization header containing the digest response.

An outbound call to an international number fails because the prepaid SIP trunk account has insufficient balance to cover the per-minute rate.

A PBX tries to place an international call but the SIP trunk provider's account has international dialing disabled. The provider rejects the INVITE outright.

A SIP trunk rejects an INVITE because it originated from an IP address not in the provider's ACL.

A user dials extension 4500, but no extension with that number exists on the PBX.

An outbound call fails because the PBX sent the number in 10-digit format but the SIP trunk provider requires E.164 format with the +1 prefix.

A PBX sends a MESSAGE request to a SIP trunk provider that only supports voice calls, not SIP messaging.

A client sends a request with an Accept header specifying a content type the server cannot produce.

An outbound INVITE hits the SIP proxy, which requires the PBX to authenticate before it will route the call. The proxy challenges with Proxy-Authenticate instead of WWW-Authenticate.

The PBX sends an INVITE to a remote extension that went offline after its last registration. No device responds, and the transaction timer expires.

A call to a former employee's extension fails because the extension was deleted from the PBX with no forwarding destination configured.

A PUBLISH request for presence state includes an If-Match header with an ETag that no longer matches the current state on the server.

An INVITE includes an SDP body listing dozens of codec options and multiple media lines, pushing the message beyond the server's size limit.

A misconfigured PBX appends excessive URI parameters to the Request-URI, exceeding the server's length limit.

An INVITE includes a multipart MIME body with an XML attachment, but the receiving server only understands application/sdp.

A request uses a tel: URI scheme but the server only handles sip: URIs.

An emergency services gateway sends a request with a Resource-Priority value that the receiving server does not recognize.

A phone sends an INVITE with Require: 100rel, but the destination PBX does not support provisional response reliability. The Unsupported header identifies the rejected extension.

The server requires session timer support but the INVITE did not include the timer extension in its Supported header.

A PBX sends an INVITE with a 90-second session timer, but the provider requires at least 1800 seconds. The Min-SE header tells the PBX what value to use.

A phone tries to register with a 30-second expiry, but the registrar requires at least 60 seconds to avoid excessive re-registration traffic.

An E911 call includes a Geolocation header with a PIDF-LO body that contains invalid coordinates or is missing required civic address fields.

A SIP trunk provider rejects an outbound INVITE because it requires STIR/SHAKEN attestation and the PBX did not include an Identity header.

A call transfer fails because the REFER request does not include a Referred-By header identifying who initiated the transfer.

A PBX sends an INVITE with Privacy: id to hide the caller's identity, but the destination provider does not accept anonymous calls.

An INVITE includes a STIR/SHAKEN Identity header, but the URL pointing to the signing certificate is unreachable or returns an error.

The STIR/SHAKEN certificate referenced in the Identity header is from a certificate authority the verifying server does not trust.

The verifying server retrieves the STIR/SHAKEN certificate successfully, but the cryptographic signature in the Identity header does not match the call's From header and date.

A proxy attempts to fork an INVITE to a large ring group, but the number of parallel branches exceeds the proxy's configured maximum.

A PBX sends a mid-call INFO request with a DTMF payload type the far end does not support. The Info-Package header identifies the rejected type.

A relay server rejects a request because the target URI has not granted permission to receive traffic from the source address.

A desk phone was unplugged and its registration expired. The PBX knows the extension exists but has no active contact to route the call to.

After a network blip, one side sends a BYE for a call that the server already tore down due to a session timer expiry. The server has no record of the dialog.

Extension 3005 forwards to extension 2001, and extension 2001 forwards back to 3005. The PBX detects the loop after the second forward.

An INVITE passes through multiple SIP proxies and the Max-Forwards counter reaches zero before the request arrives at the destination.

A user dials 7 digits but the SIP trunk provider expects 10 or 11 digits (E.164). The provider cannot route the call with an incomplete number.

A call to sip:[email protected] matches multiple users (John Smith and Jane Smith). The server cannot determine which one was intended.

An extension is on an active call and has call waiting disabled. The PBX returns 486 so the caller hears a busy signal or gets sent to voicemail.

The caller hangs up while the destination phone is still ringing. The PBX sends a CANCEL, and the callee's side responds 487 to confirm the INVITE was terminated.

One PBX is configured for SRTP-only and the other only offers RTP. The SDP negotiation fails because neither side will accept the other's media security setting.

A phone sends a SUBSCRIBE for a message-waiting-indicator event, but the server does not support that event package.

Both sides of a call send a re-INVITE simultaneously (e.g., both try to put the call on hold at the same time). One side rejects with 491 so the other can retry.

An INVITE includes an S/MIME encrypted SDP body, but the receiving server does not have the private key needed to decrypt it.

A server requires a security agreement (e.g., TLS or IPsec) to be negotiated before processing requests, but the client did not include a Security-Client header.

The PBX encounters a database error while looking up the dial plan for an outbound call. It returns 500 because it cannot determine how to route the call.

A client sends an UPDATE request to a server that has not implemented that method.

An SBC routes a call to a downstream PBX, but the PBX returns a garbled SIP response that the SBC cannot parse.

A SIP trunk provider's SBC is under maintenance and immediately rejects calls, indicating when to retry via the Retry-After header.

An SBC forwards an INVITE to a downstream PBX, but the PBX never responds. The SBC's internal timer expires and it reports the timeout to the originator.

A request arrives with SIP/3.0 in the start line, but the server only implements SIP/2.0.

A SIP message accumulates dozens of Via and Record-Route headers after passing through many proxies, pushing the total message size beyond the server's UDP MTU or configured limit.

A mobile SIP client requests push notification wakeup via the pn-provider parameter, but the registrar does not support that push notification service.