[Asterisk-Users] SPEEX CODEC and Voicepulse

[Robert Goodyear]

I'm trying to use the SPEEX codec with Voicepulse.

Here's what I see in the CLI when I RELOAD:

     -- Reloading module 'codec_speex.so' (Speex/PCM16 (signed linear) 
Codec Translator)
   == Parsing '/etc/asterisk/codecs.conf': Found
     -- CODEC SPEEX: Setting Quality to 5
     -- CODEC SPEEX: Setting Complexity to 5
     -- CODEC SPEEX: Perceptual Enhancement Mode. [on]
     -- CODEC SPEEX: VAD Mode. [off]
     -- CODEC SPEEX: VBR Mode. [off]
     -- CODEC SPEEX: Setting ABR target bitrate to 12000
     -- CODEC SPEEX: Setting VBR Quality to 5
     -- CODEC SPEEX: DTX Mode. [off]


So... it appears it's compiled, working and recognized, but it fails 
when I try to initiate a call, thus:

     -- Executing SetCallerID("SIP/501-96c1", "9492690200") in new stack
     -- Executing Dial("SIP/501-96c1", 
"IAX2/********:*********@vpconnect-t01/19493886245") in new stack
     -- Called *********:*********@vpconnect-t01/19493886245
     -- Call accepted by 66.234.228.160 (format speex)
     -- Format for call is speex
     -- IAX2/vpconnect-t01/7 is circuit-busy
     -- Hungup 'IAX2/vpconnect-t01/7'

I cannot tell if Voicepulse is not accepting the SPEEX session 
(although they say it is supported) or if it's a problem on my end.

Any ideas?

Thanks,
/rg