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)
Thanks,
Hisaharu
(2012/05/30 8:44), David Rowe wrote:
> Hello Hisaharu,
>
> Good to see you got it working. Can you pls post the IAX and
> extensions.conf information for a loop back trunk? I will add it to the
> README.
>
> Thanks,
>
> David
>
> On Tue, 2012-05-29 at 10:06 +0900, Hisaharu SUZUKI wrote:
>> Hi Bruce, David,
>>
>> Thank you for your information.
>> Finally I could listen the voice of codec2 from Asterisk!
>>
>> I changed the lines in that instruction.
>> cp ../codec2-dev/asterisk/codec_codec2.c ./codecs
>> cp ../codec2-dev/asterisk/ex_codec2.h ./codecs
>>
>> and edit source for David's advice
>> vi ./codecs/codec_codec2.c
>> -tmp->codec2 = codec2_create(CODEC2_MODE_2500);
>> +tmp->codec2 = codec2_create(CODEC2_MODE_2400);
>>
>> Actually I configured the iax2 loopback trunk for internally,
>> I could hear the voice at one asterisk server.
>>
>> I am thinking to implement the codec2 for some soft phone(ex. linphone).
>> Users could talk directly with codec2.
>>
>> Thanks,
>> Hisaharu
>>
>>
>> (2012/05/29 9:11), David Rowe wrote:
>>> Hello Hisaharu,
>>>
>>> there is one possible problem.
>>>
>>> I have recently removed the 2500 bit/s mode and now have a 2400 bit/s
>>> mode. However the Asterisk code was written for 2500 bit/s so may be
>>> broken. I don't have access to two Linux machines at the moment so
>>> can't test this.
>>>
>>> I'll look into it later this week.
>>>
>>> Cheers,
>>>
>>> David
>>>
>>> On Mon, 2012-05-28 at 21:12 +0900, Hisaharu SUZUKI wrote:
>>>> Dear David and list,
>>>>
>>>> This is my first e-mail to this list.
>>>> I am Hisaharu Suzuki at Evixar Japan.
>>>> I appreciate your interesting work.
>>>>
>>>> Recently I work on Asterisk related field,
>>>> and want to follow below instruction for technical interest.
>>>> https://freetel.svn.sourceforge.net/svnroot/freetel/codec2-dev/asterisk/README
>>>>
>>>> I use this patch to asterisk 1.8.12.
>>>> The compile was succeed, but I could not use codec2 in asterisk.
>>>>
>>>> The command of "core show codecs" show the codec2 in the list, like below.
>>>>
>>>> 8589934592 (1<< 33) (0x200000000) audio speex16 (SpeeX
>>>> 16khz)
>>>> 17179869184 (1<< 34) (0x400000000) audio codec2 (Codec 2)
>>>>
>>>> But "core show translation" show the unfortunate result,
>>>> all translation path from other codecs to codec2 shows the "-".
>>>>
>>>> "ldconfig -p" shows the libcodec2 correctly, like below.
>>>> libcodec2.so.0 (libc6,x86-64) => /usr/local/lib/libcodec2.so.0
>>>> libcodec2.so (libc6,x86-64) => /usr/local/lib/libcodec2.so
>>>>
>>>> So, someone experienced like this issue?
>>>>
>>>> #
>>>> I also found a typo in that instruction.
>>>>
>>>> -cp ~/codec2-dev/asterisk/codec2_codec2.c .
>>>> +cp ~/codec2-dev/asterisk/codec_codec2.c .
>>>> in 2/ Then build Asterisk with Codec 2 support
>>>>
>>>> --
>>>> Evixar Japan, Inc.
>>>> Hisaharu SUZUKI
>>>> [email protected]
>>>>
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>>>
>>>
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