[Freeswitch-users] FreeSwitch and Yealink

[Nandy Dagondon]

Try to set verbose_sdp=true in vars.xml. I just experienced a problem with
a China phone using Broadcom chipset.

/Nandy

On Fri, Aug 9, 2013 at 1:01 AM, Ivan <ivan at c3i.bg> wrote:

> You have to set the variable in the dialplan, before bridging. You can
> also set it directly in the bridge command.
>
> More details at:
>
> http://wiki.freeswitch.org/wiki/Codec_Negotiation
>
> But the only way to really know what is going on is to enable siptrace
> on your sofia profile and see what is offered before your endpoint
> hangups (if you post your debug, please post on freeswitch's pastebin).
>
> ivan
>
>
> On 08/07/2013 08:59 AM, Paul wrote:
> > Ivan thanks for your reply.
> >
> > I'm willing to try anything :)
> >
> > Where would I set those variables? In the dialplan out to the sip trunk?
> >
> > Thanks
> >
> > Paul
> >
> > On Tue, 6 Aug, 2013 at 10:48 PM, Ivan <ivan at c3i.bg> wrote:
> >> Hangups with the cause you describe can be caused by a failed codec
> >> negotiation. Are you sure it's not the case ? Does the remote phone
> >> send a sip "not acceptable here" ? If needed you can force the
> >> selection of the "outgoing" codec:
> >> http://wiki.freeswitch.org/wiki/Channel_Variables#absolute_codec_stringivan
> >> On 08/07/2013 07:44 AM, Paul wrote:
> >>
> >>     Hi guys, Has anyone had any issues using FreeSwitch with Yealink
> >>     phones? My phones connect to FS via openvpn tunnel. All incoming
> >>     calls work no problem, call comes through phones ring everyone can
> >>     hear each other no issues, having a very strange issue though on
> >>     the outgoing calls. As soon as the destination party picks up
> >>     (this external calls) the call hangs up. short FS LOG:
> >>     switch_ivr_bridge.c:475 Channel [sofia/internal/105 at 10.0.0.34] has
> >>     been answered sofia.c:6528 Hangup sofia/internal/105 at 10.0.0.34
> >>     [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup
> >>     sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] So
> >>     FS thinks the phone sent a BYE packet (which I can see with
> >>     siptrace) but the phone's timer keeps going as if it thinks the
> >>     call is supposed to keep going. Internal extension to extension
> >>     works fine (even if the extensions are at a different physical
> >>     location and subnet). I setup a second account to one of my
> >>     asterisk servers and outgoing/incoming work just fine, so it seems
> >>     this strange combination of FS and Yealink ... does it on 2 models
> >>     T32G and T38G (only phones I have). I have updated firwmare to
> >>     their latest version (which in the comments say freeswitch ready)
> >>     Wondering if anyone else had any experience with these, or has
> >>     some thoughts? Thanks Paul
> >>
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