Codec negotiation with carriers
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>
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>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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>
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi If, upload a file to trace and explanation to this address http://pastebin.freeswitch.org/18599 I do not want disturb watching this ;-), I prefer to use a system to understand, this scenario and for future projects. With everything I've read do not really understand what the codecs :-S , but if I had been good to understand the rest of the operation of FreeSWITCH (or so I think). Thanks ! Regards Anto 2012/3/14 Michael Collins <msc@freeswitch.org>:
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>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Well, this is a little better, however you don't have proper freeswitch logs on all these calls. For example, only the first call has freeswitch debug output. The other calls have sip traces, but not the first call. One call has what appears to be info-level output, but not debug-level output. I'd recommend that if you have this much information it might be good to put each call example in its own pastebin. Also, be sure to give a detailed description of what kind of call you are documenting. Some of your traces/debugs have no information explaining what the call is doing. Whether you are reporting a working or failed call, be sure to mention what kind of call it is. In the case of a failed call, be sure to mention what it is you are trying to do and what call result you expected to see. Thanks! -MC On Thu, Mar 15, 2012 at 2:22 PM, Anto <potxoka@gmail.com> wrote:
Hi
If, upload a file to trace and explanation to this address http://pastebin.freeswitch.org/18599
I do not want disturb watching this ;-), I prefer to use a system to understand, this scenario and for future projects.
With everything I've read do not really understand what the codecs :-S , but if I had been good to understand the rest of the operation of FreeSWITCH (or so I think). Thanks !
Regards Anto
2012/3/14 Michael Collins <msc@freeswitch.org>:
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> > > >
> Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hello I do not want to bother them, nor give me the solution. I would like to learn how to configure it for myself (in fact one of the settings works), but I do not understand the functioning correctly, for so not to disturb in the future. I attached the various configurations and their sip traces, as well as the logs (debug): First scenario ( Call isn´t established ) http://pastebin.freeswitch.org/18685 Second scenario ( Call isn´t established ) http://pastebin.freeswitch.org/18686 Third scenario ( Call isn´t established ) http://pastebin.freeswitch.org/18687 Fourth scenario ( Call isn´t established ) http://pastebin.freeswitch.org/18688 Fifth scenario ( Call established ) http://pastebin.freeswitch.org/18689 Sixth scenario ( Fail transcoding ) http://pastebin.freeswitch.org/18690 General settings: vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=VB32,G7221,speex,PCMU,PCMA,BV16,G726-32,iLBC,GSM,G729,G723,AMR"/> <!--X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729,G723,AMR,VB32,G7221,speex,BV16,G726-32,iLBC,GSM"/--> <X-PRE-PROCESS cmd="set" data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,VB32,G7221,speex,BV16,G726-32,iLBC,GSM"/> ----------------------------------------------------------------------------------------------------------------------------------------------- dialplan\outbound.xml <include> <context name="internal-in"> <extension name="outbound"> <condition field="network_addr" expression="^192\.168\.20\.40$" /> <condition field="destination_number" expression="^(\d+)$"> <action application="unset" data="sip_h_X-AUTH-IP"/> <action application="set" data="inherit_codec=true"/> <!--action application="set" data="proxy_media=true"/--> <action application="set" data="hangup_after_bridge=true"/> <!--action application="answer"/--> <action application="bridge" data="sofia/gateway/carrier1-1/$1"/> </condition> </extension> </context> </include> Thanks ! Regards Anto 2012/3/16 Michael Collins <msc@freeswitch.org>:
Well, this is a little better, however you don't have proper freeswitch logs on all these calls. For example, only the first call has freeswitch debug output. The other calls have sip traces, but not the first call. One call has what appears to be info-level output, but not debug-level output.
I'd recommend that if you have this much information it might be good to put each call example in its own pastebin. Also, be sure to give a detailed description of what kind of call you are documenting. Some of your traces/debugs have no information explaining what the call is doing. Whether you are reporting a working or failed call, be sure to mention what kind of call it is. In the case of a failed call, be sure to mention what it is you are trying to do and what call result you expected to see.
Thanks!
-MC
On Thu, Mar 15, 2012 at 2:22 PM, Anto <potxoka@gmail.com> wrote:
Hi
If, upload a file to trace and explanation to this address http://pastebin.freeswitch.org/18599
I do not want disturb watching this ;-), I prefer to use a system to understand, this scenario and for future projects.
With everything I've read do not really understand what the codecs :-S , but if I had been good to understand the rest of the operation of FreeSWITCH (or so I think). Thanks !
Regards Anto
2012/3/14 Michael Collins <msc@freeswitch.org>:
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> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hello Can anyone show me your settings to compare with mine? I look for more traces, check my settings, can not understand the functioning of the profiles :-S. Searching the internet, I haven´t found examples (a supplier or system complete) of these configurations. thanks. Regards Anto 2012/3/17 Anto <potxoka@gmail.com>:
Hello
I do not want to bother them, nor give me the solution. I would like to learn how to configure it for myself (in fact one of the settings works), but I do not understand the functioning correctly, for so not to disturb in the future.
I attached the various configurations and their sip traces, as well as the logs (debug):
First scenario ( Call isn´t established ) http://pastebin.freeswitch.org/18685
Second scenario ( Call isn´t established ) http://pastebin.freeswitch.org/18686
Third scenario ( Call isn´t established ) http://pastebin.freeswitch.org/18687
Fourth scenario ( Call isn´t established ) http://pastebin.freeswitch.org/18688
Fifth scenario ( Call established ) http://pastebin.freeswitch.org/18689
Sixth scenario ( Fail transcoding ) http://pastebin.freeswitch.org/18690
General settings:
vars.xml
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=VB32,G7221,speex,PCMU,PCMA,BV16,G726-32,iLBC,GSM,G729,G723,AMR"/> <!--X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729,G723,AMR,VB32,G7221,speex,BV16,G726-32,iLBC,GSM"/--> <X-PRE-PROCESS cmd="set" data="carriers_codec_prefs=PCMU,PCMA,G729,G723,AMR,VB32,G7221,speex,BV16,G726-32,iLBC,GSM"/>
-----------------------------------------------------------------------------------------------------------------------------------------------
dialplan\outbound.xml
<include> <context name="internal-in"> <extension name="outbound"> <condition field="network_addr" expression="^192\.168\.20\.40$" /> <condition field="destination_number" expression="^(\d+)$"> <action application="unset" data="sip_h_X-AUTH-IP"/> <action application="set" data="inherit_codec=true"/> <!--action application="set" data="proxy_media=true"/--> <action application="set" data="hangup_after_bridge=true"/> <!--action application="answer"/--> <action application="bridge" data="sofia/gateway/carrier1-1/$1"/> </condition> </extension> </context> </include>
Thanks !
Regards Anto
2012/3/16 Michael Collins <msc@freeswitch.org>:
Well, this is a little better, however you don't have proper freeswitch logs on all these calls. For example, only the first call has freeswitch debug output. The other calls have sip traces, but not the first call. One call has what appears to be info-level output, but not debug-level output.
I'd recommend that if you have this much information it might be good to put each call example in its own pastebin. Also, be sure to give a detailed description of what kind of call you are documenting. Some of your traces/debugs have no information explaining what the call is doing. Whether you are reporting a working or failed call, be sure to mention what kind of call it is. In the case of a failed call, be sure to mention what it is you are trying to do and what call result you expected to see.
Thanks!
-MC
On Thu, Mar 15, 2012 at 2:22 PM, Anto <potxoka@gmail.com> wrote:
Hi
If, upload a file to trace and explanation to this address http://pastebin.freeswitch.org/18599
I do not want disturb watching this ;-), I prefer to use a system to understand, this scenario and for future projects.
With everything I've read do not really understand what the codecs :-S , but if I had been good to understand the rest of the operation of FreeSWITCH (or so I think). Thanks !
Regards Anto
2012/3/14 Michael Collins <msc@freeswitch.org>:
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> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
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Anto
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Michael Collins