This page covers the IP address and technical details you need to let FlyNumber calls through your PBX, softphone or firewall.

Everything runs through one server, sip.flynumber.com. When a FlyNumber is set to VoIP, we send its calls from that server to your SIP address. With FlyNumber SIP, your device registers to that same server and calls flow both ways.

Below you'll find where calls originate, the media port range, supported codecs, DTMF and encryption details.

Illustration of a globe with interconnected nodes and servers, representing global network connections or IP address distribution.

IP Addresses

Calls originate from a single FlyNumber server:

  • sip.flynumber.com = 138.197.42.26 (IPv4 only, no AAAA record)

Both SIP signaling and RTP audio come from this address. If your SIP server or platform filters by source IP (an ACL, "allowed addresses" list or firewall rule), allow 138.197.42.26. When it isn't allowed, calls may appear to ring but never connect.

Ports on our side:

  • 5060 UDP for unencrypted SIP
  • 5061 for SIP over TLS

When we forward a call to your SIP address, we deliver it to the port your host publishes in DNS unless you set a non-standard port in the panel. When your device registers with FlyNumber SIP, point it at sip.flynumber.com on port 5060 (UDP) or 5061 (TLS). We don't publish SRV records, so configure the host and port explicitly.

RTP Media

Audio (RTP) travels over UDP from the same address:

  • Source IP: 138.197.42.26
  • Port range: 16384 to 32768 UDP

Your softphone or PBX needs an open path for inbound RTP. One-way or missing audio almost always means this range is blocked, or a router's SIP ALG is rewriting packets. Disable SIP ALG on any router in front of your PBX.

Supported Codecs

  • G.711 µ-law (PCMU)
  • G.711 A-law (PCMA)

Those are the only two. Wideband codecs (G.722, Opus, AAC-LD) and low-bandwidth codecs (G.729, iLBC, GSM, Speex) are not supported. Restrict your client or trunk to PCMU and PCMA; leaving Opus or G.722 enabled causes negotiation failures on some clients.

DTMF Transport Methods

DTMF signaling is supported through:

  • Telephone-event RFC 4733 (formerly RFC 2833), payload type 101

SIP INFO and inband DTMF are not supported. Set your client or trunk to RFC 2833 / RFC 4733, sometimes labelled "RTP-event" or "out-of-band".

Encryption

  • TLS 1.2 or 1.3 for signaling, with SRTP for audio using SDES key exchange. DTLS-SRTP and ZRTP are not supported.
  • When a call runs over TLS, SRTP is required. A client that connects over TLS but leaves media encryption off is rejected the moment the call is placed.
  • The sip.flynumber.com certificate is issued by Let's Encrypt and renews automatically. Trust the Let's Encrypt root in your device's CA store rather than pinning an intermediate.
  • When we forward to your server over TLS, we don't reject it for a self-signed or mismatched certificate; the handshake just has to succeed. We connect anonymously and don't send SIP credentials. If your server requires authentication, use FlyNumber SIP instead.

Asterisk Configuration Examples

  • Sample SIP Configuration (sip.conf)
  • For a TLS trunk on Asterisk or FreePBX, set media_encryption=sdes as well. A TLS trunk with encryption left off registers fine and then fails every call.

The full spec sheets live in our docs: Set to VoIP technical specifications for calls we send to you, and the FlyNumber SIP technical reference for devices that register to us.

These details apply to every virtual phone number we offer, in all 90 countries on our coverage page. See pricing for monthly rates and the Rates page for per-minute calling.

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