Hello, I have an Avaya Meeting Exchange conferencing bridge on my local network, along with an Asterisk PBX server. I can dial the MX bridge directly if I use SJPhone and the sip uri, in this case 5000@<mx_ip>. My Asterisk box has incoming DIDs. I’d like the ability to route calls coming into the Asterisk box to the MX bridge. The MX bridge can support numerous calls (ports). Is there a way to configure an Asterisk extension to do a sip uri “call” (like SJPhone) into the MX bridge? Can it be set up to do multiple calls into the MX bridge?
I’m thinking it may be a better idea to create a “trunk” between the bridge and the PBX, however, I am not sure what exactly needs to be configured in order for this to work. I’ve tried making an extension with the dial parameter set tp “SIP/5000@<mx_ip>” but this doesn’t seem to work. I’ve tried adding the extension directly to my extensions_custom.conf via:
exten => 5000,1,Dial(SIP/5000@<mx_ip>)
exten => h,1,Macro(hangupcall,)but it just rings and rings and rings (do not see anything incoming @ the bridge).
Anyone have ideas of the simplest way of forwarding multiple calls to the bridge?