On Wed, Dec 28, 2016 at 6:58 PM, Jeff Chua <jeff.chua.linux@gmail.com> wrote:
> Any help is very much appreciated!
>
> Thanks,
> Jeff
>
> On Tue, Dec 27, 2016 at 4:38 PM Jeff Chua <jeff.chua.linux@gmail.com> wrote:
>>
>> On Tue, Dec 27, 2016 at 7:03 AM, Brian West <brian@freeswitch.org> wrote:
>>
>> > Any of those variables with dashes are invalid, you should look those
>> > up,
>>
>> > those are params.
>>
>>
>>
>> Ok, I set all these to false  ...
>>
>>
>>
>> # sip_profiles/internal.xml
>>
>>         <param name="disable-transcoding" value="false"/>
>>
>>         <param name="inbound-late-negotiation" value="false"/>
>>
>>         <param name="inbound-zrtp-passthru" value="false"/>
>>
>>
>>
>> # ./sip_profiles/external.xml
>>
>>         <param name="disable-transcoding" value="false"/>
>>
>>         <param name="inbound-late-negotiation" value="false"/>
>>
>>         <param name="inbound-zrtp-passthru" value="false"/>
>>
>>
>>
>> Still no sound from GSM to SIP.
>>
>>
>>
>> What else can I try?

I still can't fix the one-way gamopen audio issue on freeswitch. It's
running the latest freeswitch git version.

I'm using Huaweil USB modem. The modem works fine in with
Asterisk/chan_dongle (audio works both ways).

SIP phone calling from wifi -> to freeswitch -> to another SIP phone
on the same wfil -> audio works both ways.

SIP phone calling from wifl -> to freeswitch -> to gsmopen -> Mobile
Phone ... -> one-way audio only from SIP phone. GSM to SIP no audio.

Thanks,
Jeff

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