Vitelity has sub-accounts but not IVR style routing yet.
Other companies offer SMS but few, if any, offer a complete working solution complete with Android and iOS apps for using the service.
We are working on real-time CDR right now, the problem is the database requirements are quite insane and it has been a challenge to provide the CDR data quickly although our development system is wicked fast and hopefully will be launched soon.
Thanks a lot for the input, user feedback is essential to know what people are looking for so we can prioritize our internal development accordingly.
vitelity features available, planned, or accepted as awesome
Does vitelity offer:
* sRTP * SIP TCP * SIP TCP TLS (avoid verizon wireless torture) * SMS filtering * visual call flow (anveo, plumvoice, voxeo) * bidirectional: sms via XMPP ** XMPP OTR encapsulation or SMS OTR via XMPP * IVR entry arbitrary digits (anveo seems to allow only 0-9 whereas voipMS easily accommodates five or more digits) * $0.50 or less setup fee for $0.99 DID * iNum * SIP Broker * SIP Broker peering * free incoming SIP * additional SIP URI not containing user id [more creative routing] * scheduled dial-out [play announcement or invoke IVR] * filter incoming CNAM * route to sequentially.ring(bob,amy,todd,tim,me,youmail) ** and else conditions * route to simultaneously.ring(bob,amy,angrybear,me) ** and else conditions * youmail-esque custom voicemail announcement based on both CID# & OCN (RDNIS) * [even] ports other than 5060 * alternate network traffic routes bound to SIP [proxy] ports (viatalk) * sms not limited to 1990s "page" size (160 characters) * accept call on DID_1 log orig CID#/CNAM/OCN route to DID_2 with inflicted CID# or OCN as CID# (until there's an iOS/android app that'll show BOTH CID# & OCN for incoming calls (network or voip)) * help Piaf users configure LCR (lowest cost routing) on f/OSS budget * RDNIS in CDR * call flow logged to CDR or analog * SIP trace (sipsorcery) * voicemail/missed call notification field separate from voicemail attachment field * more IVR flow items than anveo for each platform service level * advanced ATA tweak suggestions (OBi110,obi202,obi302,spa2102) * alphanumeric DID search [to buy DID]: starts with, contains, ends with (viatalk,LESnet) * change incoming call CNAM [Uncle Bob, Crazy Sue] (viatalk) * optional routing free toll free calls * speed dial [for SIP URI]
annoyances with voipMS:
* no CNAM filtering * feature suggestions seem to be blackholed. A quick comment per would be courteous * no development timeline * $10 cnam update per DID (callcentric updates CNAM by user editable field contents) * time filter not automatically available in my timezone * no SMS filtering (yet?) * inward LNP bumps up monthly DID cost ($0.99 to $1.49) * SIP URI contains user id * virtual DID has a cost
Vitelity's SIP trunks are essentially "clean" trunks, meaning we do not have really anything in the way of call routing. Most of this is handled through a PBX platform. While we are slowly adding features like that, a hosted PBX on on-premise PBX will do the routing.
We do not offer sRTP as our up-stream providers do not offer it so once it leaves us, it would not be secure anymore.
SIP TCP is available by adding a gateway. We have a bundled package we call Lync Enable which transcodes our UDP into TCP.
We are adding an alphanumeric search function soon. We have a pretty basic one that needs some major improvement.
Simple ring group functionality will be rolled out pretty soon
Two-way SMS is handled via an XMPP server
Most of the other requests are also handled via PBX functionality that Vitelity does not support via our user portal.