[Asterisk-Users] "Cannot native bridge" on licensed G729

Andrew Furey andrew.furey at gmail.com
Mon Jul 25 02:10:04 MST 2005


Hi folks,

In an effort to save bandwidth (our 7905s run over a WAN) we've
switched from ulaw to g729a. We purchased 4 licenses from Digium (4
SIP clients, low call volume), and they seem to have been accepted:

 [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator)
  == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e
  == Found license 'G729-253D0C86' providing 4 channels
  == Found total of 4 G.729 licenses
  == Registered translator 'g729tolin' from format G729A to SLINR, cost 5
  == Registered translator 'lintog729' from format SLINR to G729A, cost 24

*CLI> show translation
         Translation times between formats (in milliseconds)
          Source Format (Rows) Destination Format(Columns)

         G723   GSM  ULAW  ALAW  G726 ADPCM SLINR LPC10 G729A SPEEX  ILBC
   G723     -     -     -     -     -     -     -     -     -     -     -
    GSM     -     -     4     5     -     -     3     -    27     -     -
   ULAW     -    11     -     1     -     -     1     -    25     -     -
   ALAW     -    12     1     -     -     -     2     -    26     -     -
   G726     -     -     -     -     -     -     -     -     -     -     -
  ADPCM     -     -     -     -     -     -     -     -     -     -     -
  SLINR     -    10     1     2     -     -     -     -    24     -     -
  LPC10     -     -     -     -     -     -     -     -     -     -     -
  G729A     -    15     6     7     -     -     5     -     -     -     -
  SPEEX     -     -     -     -     -     -     -     -     -     -     -
   ILBC     -     -     -     -     -     -     -     -     -     -     -

*CLI> sip show peer andrew
[snip]
  Codecs       : G.729A

But when we try to use more than one (such as transferring an incoming
BRI call to a second phone), when the phone answers, the transfer
fails and we get the following:

*CLI> Jul 25 16:49:25 WARNING[114695]: chan_sip.c:2820 process_sdp: No
compatible codecs!
Jul 25 16:49:28 WARNING[114695]: chan_sip.c:2820 process_sdp: No
compatible codecs!
Jul 25 16:49:28 WARNING[114695]: chan_sip.c:2820 process_sdp: No
compatible codecs!
    -- Executing Dial("SIP/andrew-89e3", "SIP/jeremy|20") in new stack
    -- Called jeremy
    -- SIP/jeremy-b7a9 is ringing
    -- SIP/jeremy-b7a9 answered SIP/andrew-89e3
    -- Attempting native bridge of SIP/andrew-89e3 and SIP/jeremy-b7a9
Jul 25 16:49:36 WARNING[851980]: rtp.c:1392 ast_rtp_bridge: codec0 =
12 is not codec1 = 256, cannot native bridge.
  == Spawn extension (default, 801, 1) exited non-zero on 'SIP/andrew-89e3'
    -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back
from 192.168.200.226
Jul 25 16:49:42 WARNING[114695]: chan_sip.c:2820 process_sdp: No
compatible codecs!


Any ideas?

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
                          -- Bill Garrett



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