Certified and Compatible systems
Flowroute is platform agnostic, enabling you to connect any SIP-compliant device to the public switched telephone network (PSTN).
- Certified 3CX, Barracuda, SNOM, Yealink, Grandstream, Patton, beroNet, Ubiquiti, Vtech, Mobydick and Sangoma.
- Compatible Flowroute customers have success using a number of PBX’s including Cisco, Avaya, Asterisk, FreeSWITCH, Mitel, and Shoretel.
Flowroute Interconnection Specifications
Dialing format: E.164
- All numbers, including domestic destination numbers, must be dialed in international E.164 format
- Leading ‘+’ is optional
- Within the SIP INVITE request line, the E.164 number is presented in the user portion of the Flowroute server host address and in the host portion of the SIP Request URI (eg. sip:+email@example.com)
Available Authentication Methods
- SIP Digest Authentication - Defined in RFC 3261 section 22
- IP-Based Authentication (recommended)
- Requires static public IP address.
- Requires prepending dial-string with your Flowroute account 8-digit Tech Prefix, delimited with either * or # in the user portion of the SIP INVITE Request URI (eg. sip:firstname.lastname@example.org)
Transmitted and received as Out-of-Band RTP-Event packets per RFC 4733 (Commonly referred to as RFC 2833 DTMF).
Headers are implemented in line with RFC 3261. Some additional advanced headers are supported as outlined in this section:
- Caller-ID Number
- Inbound - Caller-ID Number is presented as E.164 formatted in the user portion of the From and P-Asserted-Identity headers.
- Outbound- Caller-ID Number is pulled from the user portion of one of the three following headers, in order of precedence:
- Caller-ID Name (CNAM)
- Inbound - Requires CNAM Lookup to be enabled on the called number. Caller-ID Name is presented in the Generic Name parameter of From and P-Asserted-Identity headers.
- Outbound - Caller-ID Name is set via Flowroute Manager on a per-number basis using the CNAM Storage feature (Note: The following DID types do not support CNAM storage: Toll-free numbers, Canadian numbers, iNums). The destination number must support the Caller-ID Name lookup to properly display the stored value. Caller-ID Name is not transmitted in SIP signaling. Any Generic Name parameter value is ignored.
Allows setting alternate billing number from the displayed Caller-ID Number. Recommended to set P-Charge-Info to a valid local number value when setting toll-free number as outbound Caller-ID.
Provides number(s) from which a call was diverted and the diversion reason(s).
Defined in RFC 4028. Allow for establishing a call expiration interval to avoid runaway calls if a connection is lost. Optional.
ISUP-OLI & JIP
Where available, can provide additional calling party info.
Allow logging up to 32 character strings in your call CDRs via an X-Tag header in either initial INVITE or answering 200 OK.
Codecs Supported (SDP media attribute):
- G.711 Ulaw (0)
- G.711 Alaw (8)
- DTMF tones (101)
- G.729a (18)
Recommended fax transmission guidelines
- Session negotiation: Switch to T.38 is dependent on end-user system reINVITE during an established G.711 audio codec session.
- Fallback: End-user system should support fallback to G.711 ulaw pass-through in the event of T.38 negotiation failure.
- Maximum transmission rate: 9600 baud.
- Fax Error Correction Mode (ECM): Not supported
- T.38 UDP Error Correction: T.38 UDP Redundancy
NAPTR and SRV domain support
NAPTR and SRV domain records can be used as DID routes to allow load balancing and failover on multiple servers.
IP v4 Transport Protocol Support:
SIP over UDP or TCP
Supported SIP Methods:
INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS
Unsupported SIP Methods:
PRACK, SUBSCRIBE, NOTIFY, PUBLISH, INFO, REFER, MESSAGE, UPDATE
Sent and accepted according to RFC 3261 section 21 SDP Support: RFC 4566
Flowroute stays out of the way of your call audio. We deliver the call audio stream as directly as possible to the destination carrier’s network, leveraging proprietary technology and our vast list of interconnect partners. This approach to outbound call delivery reduces latency and improves overall call quality. Read more in this KB article.