<div dir="ltr"><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span style="font-family:arial,sans-serif;font-size:13px">It logs that it is set to true</span></blockquote>
<div><br></div>Just to clarify this message, that's the 'set' application logging that the variable has been set on the channel, not anything suggesting that it is taking effect.<div><br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<span style="font-family:arial,sans-serif;font-size:13px">I just can't have a phone system with no ringing for the </span><span style="font-family:arial,sans-serif;font-size:13px">calling party.</span></blockquote><div>
<br></div><div>Is the b-leg of the bridge not sending any early media ringback? </div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On 19 September 2013 13:03, sangdrax8 <span dir="ltr"><<a href="mailto:sangdrax8@gmail.com" target="_blank">sangdrax8@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Yes, I did make sure that was activated in the profile. the<br>
configuration that comes from the git repo has that turned on, and I<br>
am using the demo config directly from git.<br>
<br>
The demo dial plan doesn't include the inherit_codec=true line, so I<br>
did add that. It logs that it is set to true, but freeswitch ends up<br>
transcoding. If I remove the ringback, suddenly everything works as<br>
expected. I just can't have a phone system with no ringing for the<br>
calling party. Still it does indicate that inbound late negotiation<br>
and the inherit_codec are capable of doing the right thing. Just when<br>
a ring back is added, it seems to fail to work.<br>
<br>
If I am configuring this correctly (only 1 change from the demo<br>
configuration provided) should I open a bug somewhere, or e-mail a dev<br>
list? I didn't want to do either of these without someone confirming<br>
that my configuration part is correct and that it is a bug.<br>
<br>
"Faithless is he, who says 'farewell', when the path darkens."<br>
"you just keep on trying till you run out of cake"<br>
<div><div class="h5"><br>
<br>
On Wed, Sep 18, 2013 at 7:31 PM, Steven Ayre <<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>> wrote:<br>
> Inherit_codec requires late negotiation - is that enabled too? Otherwise the<br>
> aleg codec has already been picked before hitting the dialplan.<br>
><br>
><br>
> On Wednesday, September 18, 2013, sangdrax8 wrote:<br>
>><br>
>> Did I explain a bit to much detail for the users list?<br>
>><br>
>> Does anyone use "inherit_codec=true" and have it working?<br>
>><br>
>><br>
>> On Tue, Sep 17, 2013 at 2:27 PM, sangdrax8 <<a href="mailto:sangdrax8@gmail.com">sangdrax8@gmail.com</a>> wrote:<br>
>> > I pulled the latest freeswitch from git this morning, and am still<br>
>> > having issues using inherit_codec=true. It seems the only way I can<br>
>> > get it to defer to the choice of the b-leg call is to remove the<br>
>> > ringback, and thus the need for early media.<br>
>> ><br>
>> > My setup has 2 phones with the following codecs configured:<br>
>> ><br>
>> > phone1000: g722, ulaw<br>
>> > phone1001: ulaw<br>
>> ><br>
>> > My goal is to have a call initiated by phone1000 (a-leg) to defer the<br>
>> > codec choice until phone1001 (b-leg) chooses one. This would allow<br>
>> > them to talk with no transcoding required by freeswitch. With the<br>
>> > following dial plan, this fails to work.<br>
>> ><br>
>> > <extension name="Local_Extension"><br>
>> > <condition field="destination_number"<br>
>> > expression="^(10[01][0-9])$"><br>
>> > <action application="export" data="dialed_extension=$1"/><br>
>> > <action application="set" data="ringback=${us-ring}"/><br>
>> > <action application="set"<br>
>> > data="transfer_ringback=$${hold_music}"/><br>
>> > <action application="set" data="call_timeout=30"/><br>
>> > <action application="set" data="hangup_after_bridge=true"/><br>
>> > <action application="set" data="inherit_codec=true" /><br>
>> > <action application="bridge"<br>
>> > data="user/${dialed_extension}@${domain_name}"/><br>
>> > </condition><br>
>> > </extension><br>
>> ><br>
>> > It does log that it is going to defer:<br>
>> ><br>
>> > 2013-09-17 13:15:10.956535 [DEBUG] mod_sofia.c:4542 [zrtp_passthru]<br>
>> > Setting a-leg inherit_codec=true<br>
>> > 2013-09-17 13:15:10.956535 [DEBUG] mod_sofia.c:4545 [zrtp_passthru]<br>
>> > Setting b-leg<br>
>> > absolute_codec_string='G722@8000h@20i@64000b,PCMU@8000h@20i@64000b'<br>
>> ><br>
>> ><br>
>> > then it tries to provide the ringback for the a-leg, and it negotiates<br>
>> > to g722 since freeswitch supports that and it is the first match.<br>
>> > Later when the b-leg is answered, it also negotiates, this time to<br>
>> > ulaw as that is all b supports. My call completes, but freeswitch has<br>
>> > to transcode between the two.<br>
>> ><br>
>> > In an attempt to see inherit_codec work, I then commented out the<br>
>> > ringback line from the above dialplan. With only this one change,<br>
>> > freeswitch now waits until the b-leg negotiation, and passes that back<br>
>> > to the a-leg. The call completes and both handsets are using ulaw and<br>
>> > no transcoding is required. The problem being that the calling party<br>
>> > never gets a ring while waiting.<br>
>> ><br>
>> > I can't imagine that the only way to use inherit_codec=true is with<br>
>> > all ringing turned off. Can someone tell me if there is an additional<br>
>> > option I am missing? All documentation I read seems to indicate that<br>
>> > just adding "inherit_codec=true" will fix things, but even the<br>
>> > default.xml provided as an example uses ringback and therefore fails.<br>
>><br>
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</blockquote></div><br></div>