Has there been any discussion about the the effect of the changes in net neutrality to VoIP service quality.It seems to me that prioritizing streaming traffic from certain content delivery companies could have an impact on the latency for VoIP wh..
Scenario:Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to 192.168.254.1:5060) is behind a NAT, acting as a client to our ITSPs SIP server. But also, this Asterisk is server for various VoIP telephones. Acoording to Asterisks wiki, the transp..
All,I have one asterisks server and 3 client (im using voip sip client for my handphone). Ive configured sip.conf and extension.conf with 3 user different. And nothing wrong with them, i could make them to make a call too.what i want to ask is, i ..
The Asterisk Development Team is pleased to announce the second beta release of Asterisk 12.0.0. You can immediately download this release at http://downloads.asterisk.org/pub/telephony/asterisk/releasesWe strongly encourage all interested Aster..
everyone. I want to migrate an old PBX which uses the Е1-PRI from one Telecom to VOIP by transparently moving the numbers one by one. I mean that the numbers that the PBX handle must be transparently moved from one operator to another. The old connect..
im planning a migration to asterisk for a high volume IVR service(from 1000 to 1500 concurrent call)The IVR service is based only on DTMF tones so the features required is- play feature- dtmf detectionAsterisk will receive calls via VOIP (SIP with g..
This is something Ive seen with some key systems and PBXs.When the phones are on-hook, they can play music throughout the office instead of having an overhead speaker system do it.Never heard of it being done with VoIP, but figured Id ask if anyone e..
Id like to replace my current VOIP provider with an Asterisk based solution.I have some ideas I want to run by the list to see if they are possible, and get answers to a couple questions.I want to setup two Asterisk servers that are linked to each othe..
What is the best way for me to setup Fax Capability with VOIP only.I have a Asterisk Server hosted on the internet without a modem. Im using Flowroute, which is working awesome, for VOIP calls.I onlyhave a SIP Phone at home and two Printer/Scanner/..
Im running Asterisk 220.127.116.11 and am connected to the nexVortex trunking service over a DSL line solely dedicated to VOIP usage.For both incoming and outgoing faxes, Im getting a failure rate of just over 25%, and over a handful of reasons.Is it natu..