Adrian Carter
2006-Mar-14 23:16 UTC
[Asterisk-Users] invalid wav gsm frame size: 1 bytes ??
I couldn't find any specific reference to this but maybe Im missing something completly... anyways, when trying to mix a few wav files together post-recording (the -in/-out files) using a pretty vanila soxmix line, I get the error: Done Mixing OUT115-20060215-150749-1139976460.7898-out.WAV..... Mixing OUT115-20060215-155022-1139979011.8787..... /usr/bin/soxmix: invalid wav gsm frame size: 1 bytes The command line for this is as basic as it gets: $SOXMIX -t WAV $LEFT -t WAV $RIGHT -t WAV $OUT I've tried playing with -r settings etc , but I always end up with 'noise' in the file in spots and it still complains about the frame size. Converting to MP3 fails compeltly using lame, and an output from soxmix to mp3 produces either a seg fault (sig 11) or a file of random periods of 'noise' and then perfectly acceptable speech (something screwy in the bitrate???) Sox ver is: 12.17.9 Supported file formats: aiff al au auto avr cdr cvs dat vms gsm hcom la lu maud mp3 nul ossdsp prc raw sb sf sl smp sndt sph 8svx sw txw ub ul uw voc vorbis vox wav wve Any suggestions ??? -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct +61 2 6163 6162 Support 1 300 662 415 E-mail cartera@lei.net.au
Tripp Meister
2006-Mar-14 23:19 UTC
[Asterisk-Users] Can't get incoming call from POTS line...
I'm having some problems but I can't seem to find other info on this one at all. What I'm using: AAH 2.7.iso (this thing rocks) - default installation X100P card for my single line home phone X-Lite softphone What I have configured in AMP: Trunk - ZAP/g1 Incoming Route = / (only thing it does it send calls to "Digital Receptionist") Outbound Route = 0 PSTN (using Trunk Sequence ZAP/g1) Digital Receptionist Extensions 200 (ZAP) 210 (SIP) 220 (SIP) Incoming Calls - send all calls to digital recep at all times and days Problem: I can make outgoing calls just fine from X-Lite but I HAVE to have an extension 200 setup for ZAP w/o it no outgoing calls will work. When an incoming call comes in, nothing happens. There's no answer and I can't figure out how or why. BTW, I'm totally new to this (been a lurker for a while to try to figure this out and have read the WIKI which is very useful but not for this). In the flash operator panel I can see the call come in but nothing happens. Thoughts? Thanks, Tripp -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060314/e47bb593/attachment.htm