[Freeswitch-users] Call progress not being passed

Bote Man bote_radio at botecomm.com
Tue Jan 19 15:35:37 MSK 2016

Good points.


Another observation: FreeSWITCH 1.6.xx-something is the latest.


If you choose to upgrade, a careful comparison of configuration files is in order as some channel variables might have changed name, and certainly some have been added.


I wish you success.






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From: Steven Ayre
Sent: Tuesday, 19 January, 2016 04:38
Subject: Re: [Freeswitch-users] Call progress not being passed


1) Are you using ignore_early_media?

2) Are you using the bridge application, or doing an originate and bridging the two channels when it connects?



On 19 January 2016 at 08:39, David Witham <david.witham at netsip.com.au> wrote:

Hi all,


We're running the latest FREESwitch 1.4 (1.4.26) on debian7.


We have a problem where we receive a SIP 183 from our upstream provider (via a sofia gateway) but the 183 does not get forwarded to the internal leg we bridge to. When we receive the 200 OK, that gets passed through and the call is established with media.


We can add a ring_ready into the dialplan and force it to pass a 180 back to the A party but that just masks the problem.


Has anyone else come across this behaviour? I haven't found any JIRAs that match this although perhaps FS-7714 is related.





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