Demo lab

The Voxserv demo lab is built to test and demonstrate some features of open-source telephony software.

Test numbers

The following PSTN numbers are set up for anyone to use. Their primary goal is to verify that DTMF and two-way audio are functioning in your outbound calls.

These numbers play a short pre-answer greeting (Child voice, "welcome to the voice demo lab"), then answer the call and offer the folllowing IVR items:

  1. Delayed echo*: the inbound audio is sent back with 1000ms delay
  2. Immediate echo*: the inbound audio is sent back without delay
  3. Audio conference*
  4. 1004Hz test sound
  5. Streaming of Radio K, University of Minnesota
  6. Sevana PVQA demo: 15 seconds of voice input are recorded and sent for impairment analysis. The results of the last 20 calls are available here. The last 3 digits of Caller ID are screened for provacy reasons.
  7. Forwards the call via two other FreeSWITCH servers and simulates 10% packet loss on your input voice. The call is then landed on this demo IVR again, and you can hear or see the effect by choosing options 1, 2, 3, or 6. If you press number 7 again, the call is looped once more, and thus 19% of packets will be dropped. This demo is implemented together with Admingroup LLC, Ukraine.
* Adaptive jitter buffer is enabled for these applications

Country Phone number ITSP
Switzerland +41335085707 netvoip.ch
Switzerland +41325133196 sipcall.ch
UK +441900210160 UKDDI @TelNG
US +19179461000 VoiceTel
INUM +883510001398258 Zadarma.com
International Dial 59805 at any of Zadarma.com access numbers Zadarma.com

SIP Addresses

The following SIP addresses can be accessed for unauthenticated calls. The domain name demo.voxserv.net has NAPTR and SRV records, and the NAPTR record for TCP transport has a better preference than that for UDP. The domain name ndemo.voxserv.net has only the NAPTR records, also for TCP and UDP. The domain name tdemo.voxserv.net has only the NAPTR record for TCP transport. All these domains point to the same service, and can be tested for UA compatibility.

SIP URI Description Codecs accepted
sip:attendant@demo.voxserv.net The IVR menu G.711 A Law and ยต Law
sip:att_hd1@demo.voxserv.net The IVR menu in high-definition codecs OPUS, SILK@16kHz
sip:att_hd2@demo.voxserv.net The IVR menu in high-definition codecs G722, Speex@16kHz, Speex@32kHz
sip:att_hd3@demo.voxserv.net The IVR menu in high-definition codecs iSAC@16kHz, iSAC@32kHz
sip:att_nb1@demo.voxserv.net The IVR menu in narrow-band codecs SILK@8kHz, Speex@8kHz, iLBC, GSM, G.723.1