Download aplikasi font untuk android apk. I recommend iLBC for x-lite if you need a low bandwidth codec, otherwise g711.u or g711.a (ulaw/alaw) if you have good bandwidth. But it would be a waste of bandwidth since you connect your iax channel with voipms in g729. If you don't want translation and low bandwidth codec, why don't you use iLBC or GSM on both your IAX provider and your X-lite config. I’m a newbie, and still decided to go through the painstaking process of upgrading from Asterisk 1.6 to 1.8 because I thought it would make the ilbc or gsm codec work. The ilbc and gsm codecs still don’t work, and now I have 2 new problems!! Hours later and I’m 8 steps back! 1----> On all calls, the person on the other end can hear me, but I hear nothing but dead air (one-way sound) 2----> There are no more call recordings showing up! The extensions are all still setup to always record. Any help would be GREATLY appreciated!! GSM is a supported CODEC, nothing has to be done. G729 CodecIt has been in Asterisk since pre version 1.0. ILBC is not included by default, I have never used this CODEC, is there s pecific reason why the g.729 CODEC is not sufficient for low bandwidth environment? 1 way audio is a networking NAT issue, not CODEC related. The previous poster asked you to post your complete information. Since we have no idea what you did, what system you are working on and what steps you take specific advice is not possible. I updated freePBX from 2.7 to 2.8 to 2.9, and then i IMMEDIATELY upgraded Asterisk 1.6 to 1.8. (i know i should have tested feePBX 2.9 before upgrading Asterisk to isolate any potential problems). I am using SIP trunks, but everything worked 100% ok in freePBX 2.7 with Asterisk 1.6, so 1 of them must have messed with firewall settings! Do you think both problems are firewall-related, or just the one-way voice problem? I have UDP ports 5060 and 0 forwarded, and i tried forwarding ports 5061-5065 too and putting my PBX in the DMZ, but it didn’t help.
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