Closed. This question is off-topic. It is not currently accepting answers. Want to improve this question? Update the question so it’s on-topic for Server Fault. Closed 8 years ago. Improve this question For my house I just wanted to setup a caller Id system that would auto-reply on certain callers to avoid harements. Any sugestion … Read more
Closed. This question is off-topic. It is not currently accepting answers. Want to improve this question? Update the question so it’s on-topic for Server Fault. Closed 5 years ago. Improve this question Where Is Physical Wires Exist for Internet Is It Using PSTN Wires for Transferring Data? If both are different network then How Telecom … Read more
I have to setup a server to route telephone calls coming to a particular landline no to cell phone phones depending upon the availability of user. I want to know what technology call centres or telecom giants use for the same. While exploring some such options I came across following 1) http://www.addictivetips.com/ubuntu-linux-tips/yate-is-free-voip-client-for-linux-that-supports-gtalk/ 2) http://packages.ubuntu.com/dapper/meta-ul-pbx-server Let me know … Read more
I have two Asterisk servers, E1(PRI) Gateway Asterisk 1.4 with E1(PRI) card IP-PBX Asterisk 1.8 on VM Server When my e1 gateway is failed (ex. shutdown, no power…), PSTN automatically switch to analog lines but when my IP-PBX is failed I want to make the gateway to route the call back to the PSTN again … Read more
My office is planning to have an intra-office software-based intercom system. As USB phones are costly, is it possible to do them with analog phones (connecting to RJ-11) instead? The phones won’t have any connection to the PSTN network. Also, what software would you recommend? Answer If it’s a single physical location, the cheapest and … Read more
Strange issue started with our Asterisk install yesterday. When dialling out via ISDN BRI the call fails and the following error is logged: Unable to create channel of type ‘DAHDI’ (cause 34 – Circuit/channel congestion) I have restarted the server, checked the config files and confirmed with the Line Provider (BT) that there is no … Read more
I do some technical work for an Internet Radio Show/Podcast, and need to fix something that has been broken for a while. The hosts have a Skype-In number to take listener calls, and for convenience sake, I bought and paid for a toll free number for a period of time. I used to use Asterlink … Read more
I’m trying to understand SIP, and one thing that keeps confusing me is phone numbers and dialing plans vs SIP URI’s like me@domain.com. What is the difference? How do they coexist/work together? For example, do you connect using a me@domain.com URI and then send DTMF tones over that connection (I doubt it but I’m not … Read more
I’m familiar with Networking, VoIP, Asterisk and Linux… but I’ve never deployed a VoIP with PSTN. I’ve got one week to deploy VoIP + 2 PSTN numbers, I don’t mind reading like crazy but I would like to get a recommendation on what hardware to buy (not the specific model, but just the specific kind … Read more
I’m wondering if it’s possible on a VoIP system (specifically Asterisk) over SIP trunks to redirect a call back out to the PSTN to no longer be routed through the PBX. What I’d like to do is see about brokering connections between two 10 digit numbers on the PSTN, but once the connection is brokered … Read more
