Trunk
It's like a pipe but for a SIP instead of water
Last updated
It's like a pipe but for a SIP instead of water
Last updated
With a Trunk you can receive inbound calls from PSTN/mobile or SIP and you can send calls to PSTN/mobile or SIP from multiple SIP users through the single number. You can also run a single Call Flow for multiple SIP users multiplexed on a single SIP domain
There is no limit on how many concurrent calls (aka. channels) can run through a single trunk. The only limiting factor may be your internet bandwidth (for calls made on a single host)
Setup your SIP trunks at
Trunk has 3 configurable sections
Trunk IN
Defines trunk input. It can be SIP or/and PSTN/mobile
Processing
You can select a Call Flow to be executed on a trunk
Trunk OUT
Defines trunk output. You can send traffic to PSTN/mobile networks or to another SIP address
Name your trunk
For SIP input type, define a subdomain for your trunk
You can also protect your trunk with SIP authentication. Set credentials that incoming SIP traffic must present before call is accepted
For PSTN input type, select a number
Select a Call Flow to be executed on a traffic passing through your trunk. It will make effect in parallel or in series with audio flowing through this channel, depending on the nature of your actions designed in a Call Flow. For example:
If Call Flow is nonblocking action like Translate speech it will run on all audio in the call (in parallel)
If Call Flow is a blocking action like Play audio or Play text with AI it will play that audio and execute Trunk OUT section once playback is completed
Define destination for traffic passing through your trunk. It may be sent to PSTN/mobile networks or to a SIP address. You can also keep it disabled
For PSTN destination type, select a source number that will be used for making calls
For SIP destination type, define destination SIP Forward URI (domain) and port. Configure SIP authentication credentials if Forward URI is protected on remote SIP
✅ All done. You have now created elastic SIP Trunk capable of processing any number of SIP extensions
