Did you get sip traces and logs of working vs. non-working calls and put them on pastebin? Most likely there is an explanation but it will take some time and effort to figure it out.<br><br>-MC<br><br><div class="gmail_quote">
On Wed, Mar 14, 2012 at 2:18 PM, Anto <span dir="ltr"><<a href="mailto:potxoka@gmail.com">potxoka@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello<br>
<br>
I have searched previous messages in the list, I consulted the book of<br>
FreeSWITCH (which I bought over a year), wiki and so on. I still do<br>
not understand how and why in some cases I work. Also I downloaded<br>
frontend to consult your code if there was something about this, but<br>
still the same. I have several weeks with this question and I can not<br>
find it. In the end I decided to spend the gateway to Asterisk, and<br>
you at least understand its operation. Thank you very much to all :-)<br>
<br>
Best regards<br>
Anto<br>
<br>
2012/3/11 Anto <<a href="mailto:potxoka@gmail.com">potxoka@gmail.com</a>>:<br>
> Hi<br>
><br>
> I still do not find the solution and not really understanding, because<br>
> it works:-S<br>
><br>
> regards<br>
> anto<br>
><br>
> 2012/3/7 Anto <<a href="mailto:potxoka@gmail.com">potxoka@gmail.com</a>>:<br>
>> Hello<br>
>><br>
>> Attached file, with the traces of the different tests (with different<br>
>> configurations).<br>
>><br>
>> <a href="http://pastebin.freeswitch.org/18599" target="_blank">http://pastebin.freeswitch.org/18599</a><br>
>><br>
>> I have read the url that you mentioned, the initial guide FreeSWITCH,<br>
>> that of mod_sofia, applications, etc.. I believe that most of the wiki<br>
>> (maybe when do not give the solution, read as much documentation is<br>
>> worse idea :-S, lock me more).<br>
>><br>
>> I made a configuration that works (I have not tested the audio), but<br>
>> earlier (before I started "touch" the profiles) if I could talk to a<br>
>> physical phone (several times). The problem is that I can not<br>
>> understand why it works and sometimes not, and I would like to learn<br>
>> :-). Not only do and forget, so I would like to learn and less<br>
>> disturbing to the mail list and (maybe in the future) to help other<br>
>> newbies like me :-). Thanks !<br>
>><br>
>> Best regards<br>
>> Anto<br>
>><br>
>> 2012/3/7 Michael Collins <<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>>:<br>
>>> You may want to read up on codec negotiation:<br>
>>> <a href="http://wiki.freeswitch.org/wiki/Codec_negotiation" target="_blank">http://wiki.freeswitch.org/wiki/Codec_negotiation</a><br>
>>><br>
>>> There are different ways to handle codecs depending on your needs. I'd read<br>
>>> that page first and then try out some of the suggestions. If you're still<br>
>>> having trouble then I'd recommend getting SIP traces of the traffic and<br>
>>> putting them on <a href="http://pastebin.freeswitch.org" target="_blank">pastebin.freeswitch.org</a>. The gang here is pretty good at<br>
>>> looking over logs and helping with diagnosing problems. :)<br>
>>><br>
>>> -MC<br>
>>><br>
>>> On Tue, Mar 6, 2012 at 2:30 PM, Anto <<a href="mailto:potxoka@gmail.com">potxoka@gmail.com</a>> wrote:<br>
>>>><br>
>>>> Hi<br>
>>>><br>
>>>> I am following the steps in this direction<br>
>>>> "<a href="http://wiki.freeswitch.org/wiki/SBC_Setup" target="_blank">http://wiki.freeswitch.org/wiki/SBC_Setup</a>" and<br>
>>>> "<a href="http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice" target="_blank">http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice</a>",<br>
>>>> I reread the whole entire wiki (or so I lack), but do not quite<br>
>>>> assimilate or finding the right formula to operate the bridge :-S.<br>
>>>><br>
>>>> I captured traffic with ngrep, I enabled sip-trace, console logconsole<br>
>>>> 8, etc., but unless the transcoding error (only two of the hundreds of<br>
>>>> combinations of settings that I have), I have not seen anything<br>
>>>> "weird" :-S<br>
>>>><br>
>>>> I have 3 suppliers, each with this codec:<br>
>>>><br>
>>>> 1) 2) 3)<br>
>>>> G729 G729 G729<br>
>>>> G711u G711A G711A<br>
>>>> G711A G711u G711u<br>
>>>> G723 G723<br>
>>>> G722<br>
>>>> GSM<br>
>>>><br>
>>>> I think I understand that when making an outside call, FreeSWITCH<br>
>>>> follow these steps:<br>
>>>><br>
>>>> USER -> ( Dialplan -> profile (internal) -> bridge (external) -><br>
>>>> profile (external) ) -> PROVIDER<br>
>>>><br>
>>>> PROVIDER -> ( Dialplan -> profile (external) -> bridge (internal) -><br>
>>>> profile (internal) ) -> USER<br>
>>>><br>
>>>> right?<br>
>>>><br>
>>>> Internal and external I set as follows (and not many changes have<br>
>>>> done, and not remember it, because I've been testing days). If<br>
>>>> outbound (outbound-codec-prefs) all codecs specified system does not<br>
>>>> handle the call, I have to specify these by hand. If active<br>
>>>> inbound-proxy-media, not the caller. Some of the time I worked, but<br>
>>>> gave me an error that it can do transcoding G729 codec (I do<br>
>>>> passthrough), but the proxy does not work half.<br>
>>>><br>
>>>> If the outbound property (outbound-codec-prefs) all codecs specified<br>
>>>> system does not handle the call, I have to specify these by hand. If<br>
>>>> active inbound-proxy-media, not the caller. Some of the time I worked,<br>
>>>> but gave me an error that it can do transcoding G729 codec (I want to<br>
>>>> make passthrough), but the "proxy media" does not work.<br>
>>>><br>
>>>> Basically, what I do is that local users can use all the codecs<br>
>>>> allowed (iLBC, GSM, ...) and make an outside call, use the carrier<br>
>>>> that will indicate the priority but the free codec.<br>
>>>><br>
>>>> With this configuration works for me, but I would like to understand<br>
>>>> why so if it works and otherwise no. Coming to understand how to<br>
>>>> configure properly and so as not to disturb the mail list ;-). Thanks<br>
>>>> !<br>
>>>><br>
>>>> Best regards<br>
>>>> Anto<br>
>>>><br>
>>>> vars.xml<br>
>>>><br>
>>>> <X-PRE-PROCESS cmd="set"<br>
>>>><br>
>>>> data="global_codec_prefs=iLBC,G7221,speex,PCMU,PCMA,BV16,G726-32,GSM,G729,G723,AMR"/><br>
>>>> <X-PRE-PROCESS cmd="set"<br>
>>>><br>
>>>> data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,iLBC,G7221,speex,BV16,G726-32,GSM"/><br>
>>>><br>
>>>> internal.xml<br>
>>>><br>
>>>> <param name="codec-prefs" value="$${global_codec_prefs}"/><br>
>>>> <!--param name="inbound-codec-prefs" value="$${global_codec_prefs}"/><br>
>>>> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/--><br>
>>>> <!--param name="inbound-codec-negotiation" value="generous"/--><br>
>>>> <param name="inbound-codec-negotiation" value="greedy"/><br>
>>>> <param name="inbound-late-negotiation" value="true"/><br>
>>>> <param name="renegotiate-codec-on-reinvite" value="true"/><br>
>>>> <!--param name="inbound-proxy-media" value="true"/--><br>
>>>><br>
>>>> external.xml<br>
>>>><br>
>>>> <param name="codec-prefs" value="$${carriers_codec_prefs}"/><br>
>>>> <param name="inbound-codec-prefs" value="$${carriers_codec_prefs}"/><br>
>>>> <!--param name="outbound-codec-prefs" value="$${carriers_codec_prefs}"/--><br>
>>>> <param name="outbound-codec-prefs" value="PCMU,PCMA,G729,G723"/><br>
>>>> <!--param name="inbound-codec-negotiation" value="generous"/--><br>
>>>> <param name="inbound-codec-negotiation" value="greedy"/><br>
>>>> <param name="inbound-late-negotiation" value="true"/><br>
>>>> <param name="renegotiate-codec-on-reinvite" value="true"/><br>
>>>> <!--param name="inbound-proxy-media" value="true"/--><br>
>>>><br>
>>>> dialplan/outbound.xml<br>
>>>><br>
>>>> <include><br>
>>>> <context name="internal"><br>
>>>> <extension name="outbound"><br>
>>>> <condition field="destination_number"<br>
>>>> expression="^(\d+)$"><br>
>>>> <!--action application="set"<br>
>>>> data="inherit_codec=true"/--><br>
>>>> <!--action application="set"<br>
>>>> data="hangup_after_bridge=true"/--><br>
>>>> <!--action application="set"<br>
>>>> data="proxy_media=true"/--><br>
>>>> <action application="bridge"<br>
>>>> data="sofia/gateway/provider-2/$1"/><br>
>>>> </condition><br>
>>>> </extension><br>
>>>> </context><br>
>>>><br>
>>>> _________________________________________________________________________<br>
>>>> Professional FreeSWITCH Consulting Services:<br>
>>>> <a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
>>>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
>>>><br>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
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>>>><br>
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>>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
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>>>><br>
>>>> FreeSWITCH-users mailing list<br>
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>>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>>><br>
>>><br>
>>><br>
>>> _________________________________________________________________________<br>
>>> Professional FreeSWITCH Consulting Services:<br>
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>>><br>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
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>>><br>
>>> Official FreeSWITCH Sites<br>
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>>><br>
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>>><br>
<br>
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</blockquote></div><br>