[asterisk-users] Change codec when dial from SIP to DAHDI

[d tbsky]

hi:
     forgot to mention. not only dialout DAHDI, even I dialout SIP
TRUNK, the situation is the same:

      asterisk transcode in the middle even two legs use the same code.

2014-09-25 11:20 GMT+08:00 d tbsky <tbskyd at gmail.com>:
> hi:
>     I Have tried asterisk 1.6.2, 1.8, 11, 12, 13. all versions behave
> the same => transcode in the middle even two legs use the same code.
>
>  but I found an article which seems to solve this kind of problem:
>
>     https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
>
>    but I tried version 13 and didn't notice the change, are there new
> diaplan commands or channel variables to do this?
>
>    thanks a lot for help!!
>
> Regards,
> tbskyd
>
>
>
>
> 2014-09-24 1:30 GMT+08:00 d tbsky <tbskyd at gmail.com>:
>> Hi:
>>      I am useing asterisk 11.12.
>>      I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI
>> use alaw. G722 is great when ip-phone talks to each other. but when
>> ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to
>> transcode to alaw.
>>      so I try to change the codec when dial from SIP to DAHDI. I tried
>> to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP codec
>> change after dahdi answered the channel. so everything is broken. the
>> call log like below:
>>
>>      [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c:     --
>> Executing [s at macro-dialout-trunk-predial-hook:2]
>> Set("SIP/222-00000004", "SIP_CODEC=alaw") in new stack
>> [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c:     --
>> Executing [s at macro-dialout-trunk-predial-hook:3]
>> Set("SIP/222-00000004", "SIP_OUT_CODEC=alaw") in new stack
>> [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c:     --
>> Executing [s at macro-dialout-trunk-predial-hook:4]
>> MacroExit("SIP/222-00000004", "") in new stack
>> [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c:     --
>> Executing [s at macro-dialout-trunk:18] GotoIf("SIP/222-00000004",
>> "0?bypass,1") in new stack
>> [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c:     --
>> Executing [s at macro-dialout-trunk:19] ExecIf("SIP/222-00000004",
>> "1?Set(CONNECTEDLINE(num,i)=0912345678)") in new stack
>> [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c:     --
>> Executing [s at macro-dialout-trunk:20] ExecIf("SIP/222-00000004",
>> "1?Set(CONNECTEDLINE(name,i)=CID:222)") in new stack
>> [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c:     --
>> Executing [s at macro-dialout-trunk:21] GotoIf("SIP/222-00000004",
>> "0?customtrunk") in new stack
>> [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c:     --
>> Executing [s at macro-dialout-trunk:22] Dial("SIP/222-00000004",
>> "DAHDI/g2/0912345678,300,Tt") in new stack
>> [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] app_dial.c:     --
>> Called DAHDI/g2/0912345678
>> [2014-09-23 21:18:53] VERBOSE[11634][C-0000000d] app_dial.c:     --
>> DAHDI/2-1 answered SIP/222-00000004
>> [2014-09-23 21:18:53] NOTICE[11634][C-0000000d] chan_sip.c: Changing
>> codec to 'alaw' for this call because of ${SIP_CODEC} variable
>> [2014-09-23 21:18:53] NOTICE[11634][C-0000000d] chan_sip.c: Changing
>> codec to 'alaw' for this call because of ${SIP_CODEC} variable
>>
>>    if I check channel with "core show channel xxxxx", got DAHDI/SIP
>> legs final like this:
>>   NativeFormats: (alaw)
>>   WriteFormat: slin
>>   ReadFormat: slin
>>   WriteTranscode: Yes (slin)->(alaw)
>>   ReadTranscode: Yes (alaw)->(slin)
>>
>>   although two legs finally use alaw both, but transcode use slin in
>> the middle. is it possible to prevent the transcode?
>>
>>   if that is not possible, then maybe I should give up using G722 as
>> the preffered codec of ip phone. back to G711 seems much  easier to
>> make all legs with the same codec.
>>
>>    thanks a lot for help!!
>>
>> Regards,
>> tbskyd