I have call queue management system where all call comes in, put in the queue while the caller speak with the online support team / teacher. However, my major concern is those under MOH (in the queue) will not be able to listen to the teacher until their turns and this is required.
my Request, I want to make the teachers audio as the MOH and every new calls that comes in listen to the life channels with the teachers.
Any help will be highly welcome.
We've got a client with a Polycom IP331 on Virgin Media behind a VMDG280 Router. Asterisk is in a data centre with a public IP.
According to the link below, there seems to be a fault with the router that blocks all SIP but the client doesn't want to change it.
Looking at the SIP DEBUG on the server, Asterisk receives the packets from the IP331, but the Polycom doesn't receive anything back from Asterisk. Port forwarding doesnt seem to help and theres no SIP ALG options on the webadmin of the router.
Has anyone successfully connected a SIP endpoint (Polycom IP331) with…
I would like to know, anyone who worked in Email to Fax scenario? If so please share the idea for implementing it.
As on other hand I configured Asterisk for inbound Fax which is working good i.e. later forward the fax via email but don't know how can I implement for outbound fax in this case.
I ran into a case today where using the "Console/DSP" that the SIP channel had been hung up a long time ago. I tried to call back into the Console/Dsp and I got busy. There was no active channel any longer. Some how it did not get the hangup.
I am running 1.4.43
How can I make sure my console channel will always - always hangup if there is no channel?