Free SIP Trunk to the PSTN for your lab

If you have a home voice lab like I do, I’m sure that you have said “I wish I had PSTN connectivity” at one point or another. My problem has been that I don’t need PSTN connectivity often enough to justify an ongoing expense, so I’m always on the hunt for something free. I used to have a free 2-channel SIP trunk from IPComms.net but it only allowed inbound calls. IPComms eventually discontinued the free Asterisk-based trunks that they had been previously offering and started removing accounts that had not been used for some time, which caused me to lose my trunk.

Recently I needed to work on something that required me to out dial from CVP, so I went on a mission to find something low cost. During my search, I found a free service over at sip-ua.com that allows outbound calls to the US for no cost and allows inbound calls via an extension based auto-attendant.

The only catch is that outbound calls are limited in duration to two minutes, which is no problem for what I’m using it for. They offer a sample CUBE config for Cisco gateways that was straight forward and very clean. I had it up and running on my 3725 in just a few minutes. I have included my config below for those that may be interested. You can also check out the sip-ua.com sample config here. One thing to note from my config versus the sample they provided is that for inbound calls, you have to add the destination dial-peer (999 in my config) to send the inside SIP leg to the destination phone or switch; they don’t show this in the sample.

Sign up for a trunk here.

If there is another service that you use or if this helped you, drop me a comment and let me know.  Thanks to our Cisco Voice brethren at sip-ua.com!

voice service voip
 allow-connections sip to sip
 sip
!
!
dial-peer voice 1 voip
 description *** 10 Digit Calls ***
 destination-pattern [2-9]..[2-9]......
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 2 voip
 description *** 11 Digit Calls ***
 destination-pattern 1[2-9]..[2-9]......
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 3 voip
 description *** Lab Extensions ***
 destination-pattern 7......
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 100 voip
 description *** Incoming Dial-Peer ***
 session protocol sipv2
 incoming called-number .
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 999 voip
 destination-pattern 4175209020
 session protocol sipv2
 session target ipv4: <ip address of phone or UCM>
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
!
sip-ua
 credentials username <username> password <password> realm sip-ua.com
 authentication username <username> password <password> realm sip-ua.com
 registrar dns:proxy.sip-ua.com expires 60
 sip-server dns:proxy.sip-ua.com
!
5 Comments
  1. Josh June 3, 2014 Reply
  2. Joe March 27, 2017 Reply
    • Michael Aossey June 24, 2017 Reply
  3. Rath April 6, 2017 Reply
    • Michael Aossey June 24, 2017 Reply

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