On Tue, Dec 27, 2016 at 2:14 AM, Jeff Chua <jeff.chua.linux@gmail.com> wrote:

On Tue, Dec 27, 2016 at 1:55 AM Francis <sms@icefire.qza.net.au> wrote:

...

What model phone are you using?

Tried with iPhone SE and Nokia X3-02. Again, even failed with the 5-sec delayed echo test.

I've tried SIP-SIP calling using LinPhone (Windows) to LinPhone (iPhone) and that works with both-way voice.

Here's my codecs in vars.xml <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=L16,99,G729,GSM@40i,PCMU,PCMA"/> Thanks, Jeff

On Tue, Dec 27, 2016 at 2:24 AM Francis <sms@icefire.qza.net.au> wrote:

Am I correct in assuming you are connecting the phone via USB and configuring an interface for gsmopen?

Yes! The modem is the Huawei E153. I've tried the E1762/K3765 as well.

I've tested these modem under Windows and they works with both-way audio.

Thanks,

Jeff

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? Thanks, Jeff

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

On Wed, Sep 20, 2017 at 6:44 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:

try to change the interface devices that are used by gsopen, many times modem (dongle) vendor are messing with them. Rerea the documentation, and check out how to change those devices, eg, in various permutations. This is the only thing that comes to my mind.

This is what I used ... <param name="controldevice_name" value="/dev/ttyUSB2"/> <param name="controldevice_audio_name" value="/dev/ttyUSB1"/> And in asterisk ... audio=/dev/ttyUSB1 ; tty port for audio connection; data=/dev/ttyUSB2 ; tty port for AT commands; There are only 3 tty's listed for this device (0 1 2). Again, asterisk works on the same system using the same device.

If this do not work, you will need to debug it deeply.

Looks like I an out of options, so will start the debug. How to start this? Thanks, Jeff

On 24 September 2017 at 18:44, Jeff Chua <jeff.chua.linux@gmail.com> wrote:

On Wed, Sep 20, 2017 at 6:44 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:

...

try to change the interface devices that are used by gsopen, many times modem (dongle) vendor are messing with them. Rerea the documentation, and check out how to change those devices, eg, in various permutations. This is the only thing that comes to my mind. If this do not work, you will need to debug it deeply.

Giovanni,

I kept getting this message ... could this be causing the no-sound from cellphone to freeswitch?

2017-09-25 00:21:48.187031 [DEBUG] mod_gsmopen.cpp:688 rev 1.9.0~64bit [(nil)|37 ][DEBUG_GSMOPEN 688 ][gsm01 ][-1, 5, 5] read more than 320, samples=640

No idea. The only suggestion/hint comes to my mind is: check if during initialization, and then when calls are made or received, it uses exactly the same AT commands and arguments as in the Asterisk channel you saw is working with your modem. If not, patch it to do the same when your modem model is detected, and then post a pull request, or send me a patch.

Thanks, Jeff.

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