[PLUG] SIP Providers with Asterisk/FreePBX

Scott Bigelow epheph at gmail.com
Wed Aug 22 00:28:12 UTC 2012


There seems to be quite a bit of Asterisk and phone expertise in the
meetings so I thought I'd reach out to the group to see if my SIP provider
is being unreasonable. I have an outbound SIP trunk configured with our
provider, nexVortex. On this SIP trunk, I have allowed the g729 codec and
disabled all others. There are a couple numbers that when we call out on
this trunk, give us a "488 Not acceptable here", usually a codec error.
Working with our provider, the common thread is that those numbers are all
hosted by nexVortex themselves, and not hitting the PSTN. They are simply
proxying the connection to the other customer's phone server.

Am I unreasonable in thinking that it is nexVortex's job to transcode this
connection, and not burden us with supporting all codecs their customers
might be using? Or should I be allowing more codecs on my trunk? Thanks!



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