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[mircea.huh at gmail.com]

Hello,

I have freeswitch version 1.10.5 -release-17-25569c1631 64bit installed.

I have one SIP provider, no nat.

I use a simple dialplan which bridge the received call from SIP provider to a mobile number using the same provider.

The call goes through but I have no ringback tone and if I answer there is no audio.

If I send the call to one internal extension, it is working fine.

The only difference I can see in the logs is that, on the call with no audio, I do not have in logs „Correct ip/port confirmed”.

Any suggestion is greatly appreciated.

Thank you.

Best regards,
Mircea.

[dragos at freeswitch.org]

Check param ext-rtp-ip in the external SIP profile. 



On Wed, Mar 10, 2021 at 12:35 AM Mircea Botoca-Huh <mircea.huh@gmail.com (mircea.huh@gmail.com)> wrote:

[mircea.huh at gmail.com]

Thank you for your answer.

I have ext-rtp-ip set to correct IP address. 


Only the forwarded call is not working. The incoming to extension has audio. 


BR, 


Mircea 


mie., 10 mar. 2021, 15:05 Dragos Oancea <dragos@freeswitch.org (dragos@freeswitch.org)> a scris:

[david.villasmil.work a...]

you should send the sip traceRegards,


David Villasmilemail: david.villasmil.work@gmail.com (david.villasmil.work@gmail.com)
phone: +34669448337






On Wed, Mar 10, 2021 at 5:42 PM Mircea Botoca-Huh <mircea.huh@gmail.com (mircea.huh@gmail.com)> wrote:

[brian at freeswitch.com]

Are you saying a hair pin call thru your FreeSWITCH has no audio (i.e. call comes in and goes right back out)?

for giggles try this at fs_cli:


global_setvar execute_on_answer01='playback silence_stream://500'


What you describe sounds like both ends are in a standoff waiting on the other to send media.


/b







On Wed, Mar 10, 2021 at 11:50 AM Mircea Botoca-Huh <mircea.huh@gmail.com (mircea.huh@gmail.com)> wrote:

[mircea.huh at gmail.com]

Hello Brian,

You were absolutely right.

I followed your advise and indeed with this variable set the call has audio.

Thank you all very much.

Best regards,
Mircea.

From: Brian West (brian@freeswitch.com)
Sent: miercuri, 10 martie 2021 20:25
To: FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org)
Subject: Re: [Freeswitch-users] No audio on bridged call


Are you saying a hair pin call thru your FreeSWITCH has no audio (i.e. call comes in and goes right back out)?


for giggles try this at fs_cli:



global_setvar execute_on_answer01='playback silence_stream://500'



What you describe sounds like both ends are in a standoff waiting on the other to send media.



/b







On Wed, Mar 10, 2021 at 11:50 AM Mircea Botoca-Huh <mircea.huh@gmail.com (mircea.huh@gmail.com)> wrote:

[gregor at infomedia.si]

Good giggles, Brian 👍


On Wed, 10 Mar 2021 at 22:22, Mircea Botoca-Huh <mircea.huh@gmail.com (mircea.huh@gmail.com)> wrote:

[brian at freeswitch.com]

I think the feature you're hitting in the upstream switch is called 'RTP Clamping'.  They won't start sending media till you do, it should only be doing this on inbound or outbound calls NOT BOTH.  What I gave you was a trick to make FreeSWITCH inject 500ms of silence, you can lower this number, you can also just set it inside the {} on the outbound originate too.

/b




On Wed, Mar 10, 2021 at 3:49 PM Mircea Botoca-Huh <mircea.huh@gmail.com (mircea.huh@gmail.com)> wrote:

[mircea.huh at gmail.com]

Indeed Brian, this is what I did. I set it on outbound originate and it is working perfectly. 


Thank you for the tip. You are the best. 


BR, 
Mircea. 

joi, 11 mar. 2021, 16:10 Brian West <brian@freeswitch.com (brian@freeswitch.com)> a scris:

[david.villasmil.work a...]

This may be a dumb suggestion, but wouldn’t doing that by default with a gateway or mod_sofia parameter parameter solve this issue without having to go to the mailing list?





On Fri, 12 Mar 2021 at 12:52, Mircea Botoca-Huh <mircea.huh@gmail.com (mircea.huh@gmail.com)> wrote: