HI,
I have tried to implement Hisaharu SUZUKI
<https://www.mail-archive.com/[email protected]&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 <[email protected]> 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 <[email protected]>
> 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 <
>> [email protected]> 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 <[email protected]> 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:[email protected]
>>>
>>>
>>> ------------------------------------------------------------------------------
>>>
>>>
>>> _______________________________________________
>>> Freetel-codec2 mailing list
>>> [email protected]
>>> 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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