Hi Folks Now that we kindly have an Asterisk codec provided, it's great for testing out voice quality.
I tweaked the Elastix RPM in order to add it to our office phone server. Elastix is one of those drop in ready to go distros and other that you then live in the usual Centos RPM hell (I concede I don't really grok working with rpm...). I anyone wants the tweaked .spec file then shout - it's nasty though... What I did was to setup a "trunk" to call out and back in on the same server - this allows me to force a specific codec over that route and after that by careful choice of destination context I can dial out of the PBX as normal. I setup prefixes so that I can dial: 40xxx dials the number using Codec2 41xxx dials the number using LPC10 42xxx dials the number using G732.1 5kbit This is great for comparing quality of codecs quickly. A snippet from my iax.conf (sip.conf would be similar) looks something like: [codec2-out] username=forced-codec2 type=peer secret=something_secret host=asterisk.example.com disallow=all allow=codec2 context=from-trunk-iax2-codec2-out [forced-codec2] secret=something_secret type=user context=from-internal ;context=from-trunk allow=all Ed W ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ _______________________________________________ Freetel-codec2 mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/freetel-codec2
