Hi,

AND core show translation gives me the output:

core show translation
         Translation times between formats (in microseconds) for one second
of data
          Source Format (Rows) Destination Format (Columns)

           g723   gsm  ulaw  alaw g726aal2 adpcm  slin lpc10  g729 speex
ilbc  g726  g722 siren7 siren14 slin16  g719 speex16 codec2 testlaw
  g723     -     -     -     -        -     -     -     -     -     -
-     -     -      -       -      -     -       -      -       -
      gsm     -     -     2     2     1001     2     1  1001     -  3001
4001  1001     2      -       -      3     -    2003      -       2
     ulaw     -     2     -     1     1001     2     1  1001     -  3001
4001  1001     2      -       -      3     -    2003      -       2
     alaw     -     2     1     -     1001     2     1  1001     -  3001
4001  1001     2      -       -      3     -    2003      -       2
 g726aal2     -     2     2     2        -     2     1  1001     -  3001
4001  1001     2      -       -      3     -    2003      -       2
    adpcm     -     2     2     2     1001     -     1  1001     -  3001
4001  1001     2      -       -      3     -    2003      -       2
     slin     -     1     1     1     1000     1     -  1000     -  3000
4000  1000     1      -       -      2     -    2002      -       1
    lpc10     -  1001  1001  1001     2000  1001  1000     -     -  4000
5000  2000  1001      -       -   1002     -    3002      -    1001
     g729     -     -     -     -        -     -     -     -     -
-     -     -     -      -       -      -     -       -      -       -
    speex     -     2     2     2     1001     2     1  1001     -     -
4001  1001     2      -       -      3     -    2003      -       2
     ilbc     -     2     2     2     1001     2     1  1001     -
3001     -  1001     2      -       -      3     -    2003      -       2
     g726     -     2     2     2     1001     2     1  1001     -  3001
4001     -     2      -       -      3     -    2003      -       2
     g722     -     2     2     2     1001     2     1  1001     -  3001
4001  1001     -      -       -      1     -    2001      -       2
   siren7     -     -     -     -        -     -     -     -     -
-     -     -     -      -       -      -     -       -      -       -
  siren14     -     -     -     -        -     -     -     -     -
-     -     -     -      -       -      -     -       -      -       -
   slin16     -     3     3     3     1002     3     2  1002     -  3002
4002  1002     1      -       -      -     -    2000      -       3
     g719     -     -     -     -        -     -     -     -     -
-     -     -     -      -       -      -     -       -      -       -
  speex16     -     4     4     4     1003     4     3  1003     -  3003
4003  1003     2      -       -      1     -       -      -       4
   codec2     -     -     -     -        -     -     -     -     -
-     -     -     -      -       -      -     -       -      -       -
  testlaw     -     2     2     2     1001     2     1  1001     -  3001
4001  1001     2      -       -      3     -    2003      -       -


On Wed, Jul 23, 2014 at 10:01 AM, salik satti <silent....@gmail.com> wrote:

> HI,
>
>       I have tried to implement Hisaharu SUZUKI
> <https://www.mail-archive.com/search?l=freetel-codec2@lists.sourceforge.net&q=from:%22Hisaharu+SUZUKI%22>'s
> codec2 implementation for asterisk using IAX2 loop back scenario. Following
> this:
>
> Hi David,
>
> Sorry for late reply.
>
> Actually I configured Asterisk with FreePBX
> and the configuration files a little bit messy as technical sample.
>
> I have confirmed codec2 related configuration with only asterisk.
>
> Here is the system layout
>
>               iax2 internaltrunk(with codec2)
>                           ||
> A(6013) - SIP phone - Asterisk - SIP Phone - B(6014)
>
> A could call to B with dialing 6014 with ulaw.
> A could call to B with dialing 996013 with codec2.
>
> This layout is only for checking codec2 sound quality.
>
> The followings are the each configuration file.
>
> ----sip.conf----
> [6013]
> type=friend
> context=default
> host=dynamic
> user=6013
> secret=6013
> canreinvite=no
> callerid=6013
> disallow=all
> allow=ulaw
>
> [6014]
> type=friend
> context=default
> host=dynamic
> user=6014
> secret=6014
> canreinvite=no
> callerid=6013
> disallow=all
> allow=ulaw
>
> ----iax.conf----
> [internal]
> disallow=all
> host=176.34.37.154
> secret=internal
> type=user
> allow=codec2
> context=default
>
> [internaltrunk]
> disallow=all
> host=176.34.37.154
> username=internal
> secret=internal
> type=peer
> qualify=yes
> trunk=yes
> allow=codec2
> context=default
>
> ----extensions.conf----
> [default]
> ;
> ; By default we include the demo.  In a production system, you
> ; probably don't want to have the demo there.
> ;
> ;include => demo
>
> exten => 6013,1,Dial(SIP/6013)
> exten => 6014,1,Dial(SIP/6014)
> exten => 996013,1,Dial(IAX2/internaltrunk/6013)
> exten => 996014,1,Dial(IAX2/internaltrunk/6014)
>
>
>
> and when i tried to make call from softphone 6013 to 6014 using codec2
> dialplan its says :
>
>  == Using SIP RTP CoS mark 5
>     -- Executing [996014@default:1] Dial("SIP/6014-0000000b",
> "IAX2/internaltrunk/6014") in new stack
> [Jul 23 17:49:11] WARNING[5438]: chan_iax2.c:12187 iax2_request: Unable to
> create translator path for codec2 to ulaw on IAX2/internaltrunk-20679
>     -- Hungup 'IAX2/internaltrunk-20679'
> [Jul 23 17:49:11] WARNING[5438]: app_dial.c:2345 dial_exec_full: Unable to
> create channel of type 'IAX2' (cause 0 - Unknown)
>   == Everyone is busy/congested at this time (1:0/0/1)
>     -- Auto fallthrough, channel 'SIP/6014-0000000b' status is
> 'CHANUNAVAIL'
>
>
> and i have configured codec2 support for asterisk using this method:
>
> 1/ First install Codec 2:
>
>     david@cool:~$ svn co 
> https://freetel.svn.sourceforge.net/svnroot/freetel/codec2-dev codec2-dev
>     david@cool:~/codec2-dev$ cd codec2-dev
>     david@cool:~/codec2-dev$ ./configure && make && sudo make install
>     david@bear:~/codec2-dev$ sudo ldconfig -v
>     david@cool:~/codec2-dev$ cd ~
>
> this not worked So using cmake installed properly ..
>
>
>  2/ Then build Asterisk with Codec 2 support:
>
>     david@cool:~$ tar xvzf asterisk-1.8.9.0.tar.gz
>     david@cool:~/asterisk-1.8.9.0$ patch -p4 < 
> ~/codec2-dev/asterisk/asterisk-codec2.patch
>     david@cool:~/asterisk-1.8.9.0$ cp ~/codec2-dev/asterisk/codec_codec2.c .
>     david@cool:~/asterisk-1.8.9.0$ cp ~/codec2-dev/asterisk/ex_codec2.h 
> ./codecs
>     david@cool:~/asterisk-1.8.9.0$ ./configure && make ASTLDFLAGS=-lcodec2
>     david@cool:~/asterisk-1.8.9.0$ sudo make install
>     david@cool:~/asterisk-1.8.9.0$ sudo make samples
>
>
> and then i have not found codec2 in command core show codecs and then i
> followed as written in file:
>
> 7/ If codec_codec2.so won't load and you see "can't find codec2_create" try:
>
>     david@cool:~/asterisk-1.8.9.0$ touch codecs/codec_codec2.c
>     david@cool:~/asterisk-1.8.9.0$ make ASTLDFLAGS=-lcodec2
>     david@cool:~/asterisk-1.8.9.0$ sudo cp codecs/codec_codec2.so 
> /usr/lib/asterisk/modules
>     david@cool:~/asterisk-1.8.9.0$ sudo asterisk -vvvcn
>
> it does come up with codec but after setup while calling i have found the 
> above error as i described.
>
> Any solutions to that problem because i want to implement it on urgent basis. 
> Any help will be greatly appreciated.
>
>
> regards,
> salik
>
>
>
>
> On Wed, Jul 23, 2014 at 8:25 AM, salik satti <silent....@gmail.com> wrote:
>
>> Steve,
>>           What do you mean by this:
>>
>> You might have better luck if you use the same version of Codec 2 that
>> was originally used with Asterisk.  It sounds like your problem is caused
>> by using a new version of Codec 2 with old Asterisk integration code
>>
>>
>> And what version of codec2 i should use with what version of asterisk and
>> where i can find the version of codecs because on link of code i can find
>> find one simple code for codec2-dev which i am trying to use currently.
>>
>> Salik
>>
>>
>> On Sun, Jul 20, 2014 at 11:59 PM, salik satti <silent....@gmail.com>
>> wrote:
>>
>>> Steve,
>>>           What do you mean by this:
>>>
>>> You might have better luck if you use the same version of Codec 2 that
>>> was originally used with Asterisk.  It sounds like your problem is caused
>>> by using a new version of Codec 2 with old Asterisk integration code
>>>
>>>
>>> And what version of codec2 i should use with what version of asterisk
>>> and where i can find the version of codecs because on link of code i can
>>> find find one simple code for codec2-dev which i am trying to use currently.
>>>
>>> Salik
>>>
>>>
>>> On Fri, Jul 11, 2014 at 12:54 PM, Steve Strobel <
>>> steve.stro...@link-comm.com> wrote:
>>>
>>>> Salik,
>>>>
>>>> You might have better luck if you use the same version of Codec 2 that
>>>> was originally used with Asterisk.  It sounds like your problem is caused
>>>> by using a new version of Codec 2 with old Asterisk integration code.
>>>>
>>>> Steve
>>>>
>>>>
>>>> On Fri, Jul 11, 2014 at 1:55 AM, David Rowe <da...@rowetel.com> wrote:
>>>>
>>>>> It's an old mode that is no longer supported.  You can see the modes
>>>>> that are supported in codec2.h
>>>>>
>>>>> --
>>>> Steve Strobel
>>>> Link Communications, Inc.
>>>> 1035 Cerise Rd
>>>> Billings, MT 59101-7378
>>>> (406) 245-5002 ext 102
>>>> (406) 245-4889 (fax)
>>>> WWW: http://www.link-comm.com
>>>> MailTo:steve.stro...@link-comm.com
>>>>
>>>>
>>>> ------------------------------------------------------------------------------
>>>>
>>>>
>>>> _______________________________________________
>>>> Freetel-codec2 mailing list
>>>> Freetel-codec2@lists.sourceforge.net
>>>> https://lists.sourceforge.net/lists/listinfo/freetel-codec2
>>>>
>>>>
>>>
>>>
>>> --
>>> Be NiCe And WiN ThE HeaRts
>>>
>>
>>
>>
>> --
>> Be NiCe And WiN ThE HeaRts
>>
>
>
>
> --
> Be NiCe And WiN ThE HeaRts
>



-- 
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