Weight Based Routing is an innovative feature introduced in Flowroute's routing system, providing a departure from traditional routing practices. While traditional routing directs all traffic exclusively through the primary route until an issue arises, Weight Based Routing offers enhanced adaptability. This feature allows customers to specify the percentage of traffic allocated to both the primary and failover routes. Unlike the conventional approach where traffic shifts entirely to the failove...

This article contains a list of SIP codes that can help you diagnose problems.

Application-level gateway (ALG) (https://en.wikipedia.org/wiki/Application-level_gateway) is intended to assist users who have phones on private IPs. In many cases, it is implemented poorly and actually causes more problems than it solves. SIP ALG modifies SIP packets in unexpected ways, corrupting them and making them unreadable. This can give you unexpected behavior, such as phones not registering and incoming calls failing or even one-way audio. Disabling SIP ALG varies, depending on your r...

To ensure you receive all audio on your Flowroute calls, specific Port Forwarding/NAT policies should be put in place on your network.

When devices communicate across the Internet, they send messages to one another. These messages are similar to mail in that they have a TO address and a FROM address. The TO address is the IP address of the device it wants to send the message to, and the FRIM address is the IP address (and port number) to send any return messages. One of the rules of private IP addresses is that they should never be seen on the Internet because theoretically millions of devices could share that IP address, and ...

As attacks become more sophisticated, it is always good to make sure that your Asterisk system is as secure as possible. The following article provides base information on locking down and securing your Asterisk PBX. - To make brute-force attacks on your system more difficult by not revealing valid extensions on your system, add the following line to the appropriate *.conf file—for example, sip.conf or sip_general_custom.conf: alwaysauthreject=yes - To bind your Asterisk phone sys...

As part of our ongoing efforts to deliver reliable and high-quality voice communications, Flowroute has in operation a number of PoPs across the world. These additional PoPs have been architected to leverage modern cloud infrastructure. With this update, you can connect your physical or virtual communications infrastructure to Flowroute over SIP following the specs outlined below. - For more details on the four mechanisms you can use for outbound Flowroute Interconnection, ordered from most ...

3CX phone system now expects all providers to include the “rinstance” parameter in SIP messages. Read more in this KB article

With the additional Points of Presence (PoPs), we have added page elements to the Flowroute Interconnection Settings for you to take advantage of their failover and load balancing capabilities. You can now select your preferred PoP in a simple format. Preferred PoP for your Inbound Route For inbound voice routes with "Host" or "URI" as the route type, you have the option to select an alternative edge strategy for a specific route. Here you will find the location of all needed routing formats...

If your phone system has a Static IP address[1] (#note1), we recommend using IP-based authentication for your outbound call authorization. Before you begin Have the following information ready before setting up IP-based authentication: - Your Flowroute Tech Prefix found on your Account Profile (https://manage.flowroute.com/accounts/profile/) page. - A list of IP addresses prepared to set up IP-based authentication To set up IP-based authentication: - On your PBX phon...

A fairly simple IP Tables script for allowing Flowroute, SSH and HTTPS access to your system can be implemented as follows: - To allow SIP Signaling from Flowroute's Primary and Secondary proxies are as follows: iptables -A INPUT -s sip-la1.flowroute.com -p UDP --dport 5060 -j ACCEPT `iptables -A INPUT -s sip-lv1.flowroute.com -p UDP --dport 5060 -j ACCEPT - To allow RTP Media (audio) into your system (verify the RTP range your system needs as this may vary): iptables -A INPUT -p U...

Failover can be set to one of the following: - an external phone number (Failover will not work with another Flowroute DID) - a host IP with another PBX attached - SIP URI routes A failover route occurs after a number of tries to the primary DID route. When the route informs Flowroute of interruption, or failure to reach the destination message, it re-routes the signal to the failover. The caller will not see any change in the call, they will be connected smoothly to the failover ro...

Two specific policies must be put in place on your network's firewall to ensure you receive all audio on Flowroute calls: - SIP signaling (call control): Allow UDP and TCP traffic over port 5060[1] (#note1) from the following server IP addresses: Point of Presence (PoP) SIP Proxy IP Range ------ ------ UNITED STATES ------ US-West-OR us-west-or.sip.flowroute.com 34.210.91.112/28 US-East-VA us-east-va.sip.flowroute.com 34.226.36.32/28 ------ ------ AS...

On occasion configured PBX systems may need to have internal configurations changed due to service updates. When a change occurs on an edge proxy (Point of Presence) there are a few details to consider to continue service uninterrupted. Resources: - Edge Proxy information - IPs of new Edge Proxies (https://manage.flowroute.com/accounts/interconnection/setup/) - Access to local/remote PBX configurations - Access to firewall configurations - Access to the Flowroute Manage p...

The attached document describes the configuration setup between Skype for Business Server and Flowroute SIP Trunk using an AudioCodes Mediant E-SBC

Documentation for helping you set up and configure your Ubiquiti UniFi VoIP system can be found at the Ubiquiti UniFi (http://help.ubnt.com/hc/en-us/categories/200321660-UniFi-VoIP/) website.

The attached document describes configuring the Epygi QX IP PBXs for Flowroute SIP trunks.

The described configuration is generic for most Zoiper VoIP softphone versions.

The following software issue was found in with the Switchvox version 6.3.2 by Digium. - Issue: Caller ID shows the Flowroute Tech Prefix. Flowroute customers using Switchvox identified a Switchvox issue where adding spaces for the Caller ID Name option caused the Flowroute Tech Prefix to show as the Caller ID in the call signaling. Status: This issue was discovered in Switchvox version 6.3.2 (81725). Solution: Remove the spaces in the Caller ID Name option. Versions previous to ver...

If we do not have setup documentation for your PBX or phone, please try to find these configuration settings on your system as most systems require the basic settings below: - SIP Authentication Username: enter here what your account shows under the SIP Credentials (https://manage.flowroute.com/accounts/interconnection/setup/) - SIP Password: enter here what your account shows under the SIP Credentials (https://manage.flowroute.com/accounts/interconnection/setup/) - SIP Registrar/Ou...

Using Flowroute as your Outbound carrier with Twilio Programmable Voice features is quick to set up; follow the steps below to start leveraging the HyperNetwork with Twilio Programmable Voice. In this case we will be using Python as the example language for placing the outbound call. The Twilio documentation covers other programming languages as well - Login to your Flowroute account and go to Interconnection > Registration to get your account SIP Registrar/Proxy, SIP Username/Auth Us...

Flowroute now offers TLS as a signaling option for customers. TLS signaling is considered an advanced configuration and should only be attempted by customers who are familiar with TLS with their PBX.

Have you ever wanted to check out your signaling? Want to try and do some personal troubleshooting? Or do you have a ticket open that requires a PCAP (packet capture)? No matter the situation, the following article describes easy steps you can follow to get your PCAP. This article describes performing a packet capture/TCP dump for both Linux-based and Windows-based systems. The default port for SIP signaling (UDP or TCP) is port 5060; port 5160 can be used as an alternative port for SIP signalin...

Let’s look at the differences between chan_sip and chan_pjsip closer and try to point all the pros and cons out of both drivers to get a better understanding of what better fits your business or personal needs. Some history Asterisk-based PBX and servers use a channel driver to communicate along VoIP protocols. Asterisk, at its core, is agnostic and modular; meaning that it can be adapted to different signaling protocols. It utilizes channel drivers to communicate between the core and the o...

Using Flowroute as your Inbound carrier with Twilio Programmable Voice features is quick to set up; follow the steps below to start leveraging the HyperNetwork with Twilio Programmable Voice. - Login to your Twilio account and go to Programmable Voice > SIP Domains - Click IP Access Control Lists on the left under SIP Domains - Add a new list and include the following IPs ------------------------- Point of Presence (PoP) SIP Proxy IP Range ------ ------ UNITED STA...

To receive inbound calls on your FreePBX system when your Flowroute Direct Inward Dial (DID) is dialed, you must have an inbound route configured. This is required in addition to ensuring your DID points to a valid route within Flowroute Manage. By default all DIDs route to any active SIP registration. See Statically Route Your Number to a Host System for Inbound Calls (https://support.flowroute.com/733023-Statically-route-your-phone-number-to-a-host-system-for-inbound-calls) if you are not usin...