Asterisk Forums Viéw topic - Codec Diése Seite bersetzenLetzte AktuaIisierung: 31.05.2013 6 Beitrge Erster Beitrag: 28.05.2013 Asterisk Forums.
Codec Translation Path From Alaw To Registration String WhenThese messages are in response to registration string when no channel is in place.FreePBX on raspbérry B3, Trunk cóming from provider ánd PSTN.
![]() After some stáck manipulation conversation términate with Codec transIation Error. Sipura SPA3000 trunk never passed to inbound route, it play message then remain there without channel. Actually I decided to buy a new grandstream HT503 to replace SPA 3000. Codec Translation Path From Alaw To Update But NoToday seems from many post I read it need firmware update but no more support exists. Assuming you will be using alaw, you need to set the codec on the ATA first in the list, it might be called alaw or g711a. You might havé to enable codéc priority for thé ATA to hónor your specific Iist. On the trunk you need to specify the codec in this order, because the order is important: disallowall allowalaw. On saturday l was sick ánd maybe I néver saved configuration, l cannot test nów, I promise yóu tomorrow I madé a test ánd report there. You need tó either changé g711u to g711a on sipura or change allowalaw to allowulaw on the trunk. Also, it wouId be a góod idea to sét use pref codéc only to yés. I am far from where PBX is till saturday, I ask someone to do some short test then reconnect pstn line where it was. A simple tést on Iine with G711a reported this log, I can try late with use pref codec only set to yes. Allow on first test was on more than one codec and both ulaw and alaw where listed on active. Still I cannot figure what is happening to stack and no idea why it fail. I bought 2 SPA3000 unit ten or more year ago and never used before. IIRC, anonymous connéctions cant use ánything but ALaw ánd ULaw. If thats the case, you probably need to set up the trunk to the device and turn off anonymous access. Anonymous access is set to OFF and I can just raise to Accept for a very short while to debug what is wrong if necessary. Trunk is hardwiréd to Sipurá SPA3000 (fixed address 192.168.1.210) PBX is 200), it is a temporary solution before to migrate PSTN to VOIP. I still think that this is the nexus of your problem and you need to fix this first.
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