<div dir="ltr">Try to set verbose_sdp=true in vars.xml. I just experienced a problem with a China phone using Broadcom chipset.<br><div class="gmail_extra"><div><br></div>
/Nandy<br><br><div class="gmail_quote">On Fri, Aug 9, 2013 at 1:01 AM, Ivan <span dir="ltr"><<a href="mailto:ivan@c3i.bg" target="_blank">ivan@c3i.bg</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
You have to set the variable in the dialplan, before bridging. You can<br>
also set it directly in the bridge command.<br>
<br>
More details at:<br>
<br>
<a href="http://wiki.freeswitch.org/wiki/Codec_Negotiation" target="_blank">http://wiki.freeswitch.org/wiki/Codec_Negotiation</a><br>
<br>
But the only way to really know what is going on is to enable siptrace<br>
on your sofia profile and see what is offered before your endpoint<br>
hangups (if you post your debug, please post on freeswitch's pastebin).<br>
<span class="HOEnZb"><font color="#888888"><br>
ivan<br>
</font></span><div class="HOEnZb"><div class="h5"><br>
<br>
On 08/07/2013 08:59 AM, Paul wrote:<br>
> Ivan thanks for your reply.<br>
><br>
> I'm willing to try anything :)<br>
><br>
> Where would I set those variables? In the dialplan out to the sip trunk?<br>
><br>
> Thanks<br>
><br>
> Paul<br>
><br>
> On Tue, 6 Aug, 2013 at 10:48 PM, Ivan <<a href="mailto:ivan@c3i.bg">ivan@c3i.bg</a>> wrote:<br>
>> Hangups with the cause you describe can be caused by a failed codec<br>
>> negotiation. Are you sure it's not the case ? Does the remote phone<br>
>> send a sip "not acceptable here" ? If needed you can force the<br>
>> selection of the "outgoing" codec:<br>
>> <a href="http://wiki.freeswitch.org/wiki/Channel_Variables#absolute_codec_string" target="_blank">http://wiki.freeswitch.org/wiki/Channel_Variables#absolute_codec_string</a> ivan<br>
>> On 08/07/2013 07:44 AM, Paul wrote:<br>
>><br>
>> Hi guys, Has anyone had any issues using FreeSwitch with Yealink<br>
>> phones? My phones connect to FS via openvpn tunnel. All incoming<br>
>> calls work no problem, call comes through phones ring everyone can<br>
>> hear each other no issues, having a very strange issue though on<br>
>> the outgoing calls. As soon as the destination party picks up<br>
>> (this external calls) the call hangs up. short FS LOG:<br>
>> switch_ivr_bridge.c:475 Channel [sofia/internal/<a href="mailto:105@10.0.0.34">105@10.0.0.34</a>] has<br>
>> been answered sofia.c:6528 Hangup sofia/internal/<a href="mailto:105@10.0.0.34">105@10.0.0.34</a><br>
>> [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup<br>
>> sofia/external/<a href="tel:2503004900" value="+12503004900">2503004900</a> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] So<br>
>> FS thinks the phone sent a BYE packet (which I can see with<br>
>> siptrace) but the phone's timer keeps going as if it thinks the<br>
>> call is supposed to keep going. Internal extension to extension<br>
>> works fine (even if the extensions are at a different physical<br>
>> location and subnet). I setup a second account to one of my<br>
>> asterisk servers and outgoing/incoming work just fine, so it seems<br>
>> this strange combination of FS and Yealink ... does it on 2 models<br>
>> T32G and T38G (only phones I have). I have updated firwmare to<br>
>> their latest version (which in the comments say freeswitch ready)<br>
>> Wondering if anyone else had any experience with these, or has<br>
>> some thoughts? Thanks Paul<br>
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><br>
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><br>
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</div></div></blockquote></div><br></div></div>