Tag Info

New answers tagged

1

There are a couple of approaches possible. The first thing you need to understand is the concept of putting a call on "hold" while you dial a number. In practical terms Asterisk does this whenever a Dial is in progress. You can even play specific audio to tell the caller you are dialing (or on hold). Depending on what you want to do if no one answers, ...


2

Well, there is a reason Dell writes a comprehensive technical guide for these kind of systems: To answer that kind of question. See http://i.dell.com/sites/doccontent/shared-content/data-sheets/en/Documents/dell-poweredge-r510-technical-guide.pdf Spoiler: It supports only one full length card.


0

Here is an example of how I do this with google voice and trunks. With my setup I had to make a way to limit gv to 1 channel, also setup a rotation so the first gv trunk is not always hammered since it is first. There is a lot more to this but for just displaying failover I thought this was more than enough: exten => ...


1

This is caused by the use of the _. pattern. Asterisk book warns against using this pattern due to the fact it catches the special one-letter extensions.


1

Asterisk already has this in place. You just need to setup a "call queue". http://www.voip-info.org/wiki/view/Asterisk+call+queues https://www.youtube.com/watch?v=HIulFoe4r3s


2

You forgot to set localnet. The configuration file did warn you that you have to set this equal to the netblock of your local network. ;----------------------------------------- NAT SUPPORT ------------------------ ; ; WARNING: SIP operation behind a NAT is tricky and you really need ; to read and understand well the following section. ; ; When Asterisk is ...


0

Kamailio is only an SIP proxy (call negotiation), you still need a RTP server in order to handle the audio of the calls like Asterisk or FreeSwitch


1

I suspect your SIP port is now in the RTP range, so Asterisk doesn't realize this is a SIP connection. Either move the SIP port or move the RTP port range so they don't overlap. Please capture the CLI output of a failed call (off LAN) and a succesfull call (on LAN) and post the results here. To capture the CLI output, from the bash prompt: asterisk -r ...



Top 50 recent answers are included