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