[Freeswitch-users] Codec list truncation?

[Pete Ashdown]

Is there some sort of limit in my SIP rtpmap for codecs?  I've got this
list of codecs:

  <X-PRE-PROCESS cmd="set"
data="global_codec_prefs=speex at 32000h@20i,speex at 16000h@20i,speex at 8000h@20i,iLBC at 30i,G7221 at 32000h,G7221 at 16000h,opus,PCMU,PCMA,GSM"/>
  <X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=speex at 32000h@20i,speex at 16000h@20i,speex at 8000h@20i,iLBC at 30i,G7221 at 32000h,G7221 at 16000h,opus,PCMU,PCMA,GSM"/>

I'm seeing this output from tcpdump:

    v=0
    o=FreeSWITCH 1395348540 1395348541 IN IP4 10.10.10.1
    s=FreeSWITCH
    c=IN IP4 10.10.10.1
    t=0 0
    m=audio 25672 RTP/AVP 0 98 99 100 102 103 104 105 8 3 101 13
    a=rtpmap:98 SPEEX/32000
    a=rtpmap:99 SPEEX/16000
    a=rtpmap:100 SPEEX/8000
    a=rtpmap:102 iLBC/8000
    a=fmtp:102 mode=30
    a=rtpmap:103 G7221/32000
    a=fmtp:103 bitrate=48000
    a=rtpmap:104 G7221/16000
    a=fmtp:104 bitr[|sip]


Note the last line with bitr[|sip] where bitrate should be.  This causes
phones that would otherwise answer the call with the available codecs
above to ignore and not ring at all.