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