I doubt if that will work or not. Capture sip log of your call and send.
Use pastebin.com.

Best Regards

Samir Doshi
iNextrix Technologies Pvt. Ltd.
http://www.inextrix.com

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On Fri, May 26, 2017 at 12:37 PM, Dipen Patel <notifications@github.com>
wrote:

> hello..
> i have setup free switch as per your setup step.
>
> now my gateway required PCMU,PCMA codec.
>
> so i have set below setting in sip profile
> inbound-codec-prefs=GSM (that i have set in sip phone )
> outbound-codec-prefs=PCMU,PCMA
> disable-transcoding=true
>
> But still not connect GSM to PCMU,PCMA
>
> Let me know if anything is wrong by me.
> Thanks
>
> —
> You are receiving this because you are subscribed to this thread.
> Reply to this email directly, view it on GitHub
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>

ok...
thanks please see below pastbin log
Sending GSM inbound codec and want to send to gateway PCMU,PCMA

https://pastebin.com/bL1gW1RV

1. 2017-05-27 01:46:02.775480 [DEBUG] mod_sofia.c:437 Channel
sofia/default/917586911121 hanging up, cause: RECOVERY_ON_TIMER_EXPIRE
2. 2017-05-27 01:46:02.775480 [DEBUG] switch_core_state_machine.c:60
sofia/default/917586911121 Standard HANGUP, cause: RECOVERY_ON_TIMER_EXPIRE

Enable sip log using "sofia global siptrace on" command in fs_cli and
provide sip log of your call.

Best Regards

Samir Doshi
iNextrix Technologies Pvt. Ltd.
http://www.inextrix.com

Disclaimer:
The information contained in this communication is confidential and may be
legally privileged. It is intended solely for the use of the individual or
entity to whom it is addressed and others authorized to receive it. If you
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information is strictly prohibited and may be unlawful. Please notify the
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On Sat, May 27, 2017 at 11:35 AM, Dipen Patel <notifications@github.com>
wrote:

> ok...
> thanks please see below pastbin log
> Sending GSM inbound codec and want to send to gateway PCMU,PCMA
>
> https://pastebin.com/bL1gW1RV
>
> —
> You are receiving this because you commented.
> Reply to this email directly, view it on GitHub
> <https://github.com/iNextrix/ASTPP/issues/205#issuecomment-304430771>, or mute
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>

ok..
please download below log after done sofia global siptrace on
https://pastebin.com/9Q31yEM7

after call from linphone get error: Unknown Error : Request Timeout.

Re. hanging up, cause: RECOVERY_ON_TIMER_EXPIRE
i think that is because when we doing call so provider is only accepting PCMU,PCMA and we sending GSM due to some how not convert so getting that error
Thanks

Hi
When a call have two kinds of codecs you must enable transcoding.
disable-transcoding=false
By default freeswitch have inbound-late-negotiaton=true change to false this will forcé match codecs before start routing.

If nothing works you can add into gateways extra, absolutec_codec_string="PCMU,PCMA"

El 26 de mayo de 2017 2:07:18 GMT-05:00, Dipen Patel <notifications@github.com> escribió:
>hello..
>i have setup free switch as per your setup step.
>
>now my gateway required PCMU,PCMA codec.
>
>so i have set below setting in sip profile
>inbound-codec-prefs=GSM (that i have set in sip phone )
>outbound-codec-prefs=PCMU,PCMA
>disable-transcoding=true
>
>But still not connect GSM to PCMU,PCMA
>
>Let me know if anything is wrong by me.
>Thanks
>
>--
>You are receiving this because you are subscribed to this thread.
>Reply to this email directly or view it on GitHub:
>https://github.com/iNextrix/ASTPP/issues/205

–
Enviado desde mi dispositivo Android con K-9 Mail. Por favor, disculpa mi brevedad.

hello..
thanks for that info..
add into gateways extra, absolutec_codec_string="PCMU,PCMA"
i can't seen extra option in Gateway so i have added in Dialplan Variable but call not going.

i have also one more try to set "PCMU,PCMA" in trunk -> codec instead of gateway > dialplan but also not worked.

here i attached mine settings screenshot.
Please look it.