Asterisk 1.8.0, Debian Lenny
I am trying to utilize the g726 codec between an SIP endpoint and my Asterisk box. From the * box to my SIP provider I am trying to use g711u. In this case the endpoint is a Sipura-2000. The SIP provider is Callcentric. My whole configuration works flawlessly with g711u and g729a.
I read that * only supports g726-32 so I have selected that option in my Sipura ATA’s configuration. I can place a call when using the g726 codec but I cannot receive a call. When I try to route and incoming call to this Sipura endpoint while trying to use only g726 * tells me that my endpoint is unavailable. It seems like my * box is going to automatically (i think??) transcode from g726 to g711u on the outbound but how do I get it to go from g711u to g726 on the inbound?
Could this be something to do with the ATA? Is there another g726 library I should be using?