Well I am experimenting with * and am now about 2-weeks in. I would like to know how do i implement a dialplan to accept a DID if I am not on a PSTN and if it can be done at all? The DID that I have received is supposed to be forwarded to my URL (voipme2u.com), or basically my * server. I only have extensions implemented with my current dialplans in *.
Thanks for any assistance.
You have to setup a SIP or IAX connection to the asterisk box. The settings depend on your provider but basically you define the line in the sip/iax.conf file and set the context for the dial plan. In the dialplan, define a default extension and you’re off…
[quote=“tlofton1000”]thank you.[/quote] And I am able to receive the DID calls. However, I do need to work on the sound effects, i can’t hear the caller at all but the transmitting party hears me clearly (g.711 codec and sip.conf gsm, ulaw, alaw) [in case anyone wants to add something…???].
Is the asterisk server firewalled? Check to make sure you’re allowing inbound RTP traffic to the server in question (typically UDP port 10000-20000)