[Freeswitch-users] Call progress not being passed

[Steven Ayre]

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.
>
> thanks,
> David
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160119/980a113a/attachment.html