March 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Can user extension be set to use TCP?
by Mario G 24 Mar '12

24 Mar '12

I have a need to set a softphone client to use TCP. I was not able to find anything on the wiki regarding TCP for a user extention parm. The only TCP references are for gateway setttings. Does anyone know if there is a parm for the user extension to be forced to use TCP? BTW, although I have TCP set in the client and it is working, however, FS is failing every night (all internal extension fail requiring a FS restart) which is why I am hoping a setting will work. Thanks, Mario G

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Codec negotiation with carriers
by Anto 24 Mar '12

24 Mar '12

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="$${global_codec_prefs}"/>   <param name="inbound-codec-negotiation" value="greedy"/> <param name="inbound-late-negotiation" value="true"/> <param name="renegotiate-codec-on-reinvite" value="true"/>  external.xml <param name="codec-prefs" value="$${carriers_codec_prefs}"/> <param name="inbound-codec-prefs" value="$${carriers_codec_prefs}"/>  <param name="outbound-codec-prefs" value="PCMU,PCMA,G729,G723"/>  <param name="inbound-codec-negotiation" value="greedy"/> <param name="inbound-late-negotiation" value="true"/> <param name="renegotiate-codec-on-reinvite" value="true"/>  dialplan/outbound.xml <include> <context name="internal"> <extension name="outbound"> <condition field="destination_number" expression="^(\d+)$">    <action application="bridge" data="sofia/gateway/provider-2/$1"/> </condition> </extension> </context>

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INCOMPATIBLE_DESTINATION
by Anderson Arboleya 24 Mar '12

24 Mar '12

Hi, somebody knows what do I have to do in order to fix that hangup cause? Here's the complete log (with debug) of the inbound call. Spent my whole day trying to understand it.. without any luck. Thanks ============>>>>>>>>> 2012-03-24 01:18:37.865450 [NOTICE] switch_channel.c:926 New Channel sofia/external/1184049130@189.84.133.130:5060[4898b56c-754f-11e1-af7e-812af071ddc6] 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5532 Channel sofia/external/ 1184049130@189.84.133.130:5060 entering state [received][100] 2012-03-24 01:18:37.865450 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=VoipSwitch 6106 7106 IN IP4 189.84.133.130 s=VoipSIP i=Audio Session c=IN IP4 189.84.133.130 t=0 0 m=audio 6106 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[GSM:3:8000:20:13200] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2012-03-24 01:18:37.865450 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2848 (sofia/external/ 1184049130@189.84.133.130:5060) Callstate Change DOWN -> HANGUP 2012-03-24 01:18:37.865450 [NOTICE] sofia.c:5813 Hangup sofia/external/ 1184049130@189.84.133.130:5060 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2012-03-24 01:18:37.865450 [DEBUG] switch_channel.c:2871 Send signal sofia/external/1184049130@189.84.133.130:5060 [KILL] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal sofia/external/1184049130@189.84.133.130:5060 [BREAK] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1184049130@189.84.133.130:5060) Running State Change CS_HANGUP 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 (sofia/external/1184049130@189.84.133.130:5060) State HANGUP 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:469 Channel sofia/external/ 1184049130@189.84.133.130:5060 hanging up, cause: INCOMPATIBLE_DESTINATION 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:534 Responding to INVITE with: 488 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:47 sofia/external/1184049130@189.84.133.130:5060 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:602 (sofia/external/1184049130@189.84.133.130:5060) State HANGUP going to sleep 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:393 (sofia/external/1184049130@189.84.133.130:5060) State Change CS_HANGUP -> CS_REPORTING 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal sofia/external/1184049130@189.84.133.130:5060 [BREAK] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1184049130@189.84.133.130:5060) Running State Change CS_REPORTING 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 (sofia/external/1184049130@189.84.133.130:5060) State REPORTING 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:79 sofia/external/1184049130@189.84.133.130:5060 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:662 (sofia/external/1184049130@189.84.133.130:5060) State REPORTING going to sleep 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:387 (sofia/external/1184049130@189.84.133.130:5060) State Change CS_REPORTING -> CS_DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1182 Send signal sofia/external/1184049130@189.84.133.130:5060 [BREAK] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_session.c:1382 Session 5 (sofia/external/1184049130@189.84.133.130:5060) Locked, Waiting on external entities 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1400 Session 5 (sofia/external/1184049130@189.84.133.130:5060) Ended 2012-03-24 01:18:37.865450 [NOTICE] switch_core_session.c:1402 Close Channel sofia/external/1184049130@189.84.133.130:5060 [CS_DESTROY] 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:491 (sofia/external/1184049130@189.84.133.130:5060) Callstate Change HANGUP -> DOWN 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:494 (sofia/external/1184049130@189.84.133.130:5060) Running State Change CS_DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 (sofia/external/1184049130@189.84.133.130:5060) State DESTROY 2012-03-24 01:18:37.865450 [DEBUG] mod_sofia.c:374 sofia/external/ 1184049130@189.84.133.130:5060 SOFIA DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:86 sofia/external/1184049130@189.84.133.130:5060 Standard DESTROY 2012-03-24 01:18:37.865450 [DEBUG] switch_core_state_machine.c:504 (sofia/external/1184049130@189.84.133.130:5060) State DESTROY going to sleep <<<<<<<<<============

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skypopen scalability tests
by Giovanni Maruzzelli 24 Mar '12

24 Mar '12

On Sat, Apr 30, 2011 at 6:52 PM, Anton VG <anton.vazir(a)gmail.com> wrote: >> Any other ideas on how to proceed for scalability tests? >> > > Hm, since 2.0.0.72 still allows simultaneous calls, you do not have to > register lots of usernames, just 2 of them for 2 PC's > than make 100 calls from 1 machine to another I was going that way, couple years ago, but the Skype client was scaling kind of badly for me. No more than a bunch of simultaneus calls. But at that time I was testing using a standard ALSA sound driver, and Skype client for ALSA. Maybe using Skype for OSS and skypopen.ko it will scale better. Good hint, at least I will check into it in the future. Other ideas? -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Major Patch Coming this weekend
by Ken Rice 24 Mar '12

24 Mar '12

Hey Guys, In prep for 1.2 I¹ll be merging a patch that makes some adjustments to the Makefile and configure scripts, this means you¹ll have to rebootstrap. This hasn¹t been done yet, but heads up its coming.... I¹ll follow up to the list once that merge has happened as you¹ll know you whats up when your make breaks K

5 4

Push possibly breaking things
by Ken Rice 23 Mar '12

23 Mar '12

Ok that Push I meantioned is going to happen this evening... If you get breakage on make current or make after doing a pull, rebootstrap and reconfigure. K

1 0

Gateway Configuration (asterisk example provided)
by Anderson Arboleya 23 Mar '12

23 Mar '12

Hello, I'm trying to connect freeswitch to a provider so I can call real numbers and reach my freeswitch server, but I'm kinda stuck in this process. My provider sent me this Asterisk's configs: ======================================================= # sip.conf: register => USERNAME:PASSWD@SERVER_IP [directcall12] context=default fromuser= USERNAME username= USERNAME secret= PASSWD type=friend host=SERVER_IP canreinvite=no nat=no dtmfmode=rfc2833 insecure=very disallow=all allow=g729 # extensions.conf: exten => 3131,1,Noop(ORIGEM - ${CALLERID(num)}) exten => 3131,n,Dial(${TRUNKPBX}/${RAMAL}) exten => 3131,n,Hangup() ** change these settings according you internal extensions configs ======================================================= Well, I tried adapting it to FreeSwitch like this (according some examples I found across the web): ======================================================= # /usr/local/freeswitch/conf/sip_profiles/external/testing.xml <include> <gateway name="testing"> <param name="username" value="USERNAME" /> <param name="realm" value="SERVER_IP" /> <param name="password" value="PASSWORD" /> </gateway> </include> # /usr/local/freeswitch/conf/dialplan/public/testing.xml <include> <extension name="testing-inbound"> <condition field="destination_number" expression="^(USERNAME)$"> <action application="lua" data="/root/call_logger.lua" /> </condition> </extension> </include> ======================================================= In the end all I got is this log: 2012-03-17 01:41:23.565471 [NOTICE] sofia_reg.c:415 Registering testing 2012-03-17 01:41:55.565460 [ERR] sofia_reg.c:1962 testing Registration Failed with status Request Timeout [408]. All I want is to connect with my provider and route the calls made to the real number my provider gave me to my Dialplan, so my lua script will be called. Btw, I dont actually bought a "dedicated number" but an extension. I mean, I will call a number and an IVR-menu will ask me for an extension, then I type the extension I bought (3131) and their system will send the call for me. Is this clear? Sorry, I'm an experienced programmer but I'm just starting with all this telephony thing and I don't know what else I need to do from this point on, I have searched a lot and tried a lot of different approaches, but none succeed. Could anybody help me with this, please? Thanks in advance. Anderson Arboleya

4 9

Myevents
by Gregor Nanger 23 Mar '12

23 Mar '12

One explanation... If I use in outbound connection event plain <EVENTS>, I get only subscribed events. But if I also use myevents command I get all events. So, does myevents command override event plain command?

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FreeSWITCH Wiki Update: Syntax Highlighting Now Available!
by Michael Collins 23 Mar '12

23 Mar '12

Hey all, Hat tip to David Knell for figuring out how to get the GeSHi syntax highlighting to work on wiki.freeswitch.org! If you'd like to have syntax highlighting on a snippet of code just do this: <source lang="{lang}"> ...code... </source> Languages include: c, perl, python, xml, lua, etc. Example: <source lang="xml"> <extension name="test ext"> <condition field="destination_number" expression="^(1234)$"> <action application="answer"/> <action application="log" data="INFO Thanks David!!"/> <action application="hangup"/> </condition> </extension> </source> I added the basic information to this page on the wiki: http://wiki.freeswitch.org/wiki/Documentation_Guidelines#Wiki_Markup If you would like a non-glamorous but useful task to do: feel free to edit each code sample you see to have the <source> tag with the appropriate language. Also, if there are any GeSHi warriors out there who could write a FreeSWITCH Log syntax highlighter module I'm sure that would be welcomed. :) Thanks again! -Michael

1 0

ITSPA 2012 Award for Open Source VoIP Projects
by Daniel-Constantin Mierla 23 Mar '12

23 Mar '12

Hello, ITSPA UK has unveiled the winners of its 4th annual Awards, an event designed to celebrate innovation and best practice in the VoIP industry: * http://www.itspaawards.org.uk/ Open Source VoIP Projects won a special category this year, Members' Pick, for providing a real value to VoIP Industry. I had the chance to attend the event in London and I have been selected to pick up the award. I made a news on the website of the project I am mainly involved in (Kamailio) with more details: * http://www.kamailio.org/w/2012/03/itspa-awards-2012-open-source-voip-projec… As you would expect, a complete voip platform usually involves several open source projects, for components such as load balancers, registrar, proxy, gateways or media servers, thus the decision of ITSPA for awarding to the group. FreeSwitch was a frequent presence in the VoIP systems of the ITSPA members I spoke to. I guess it is no surprise at all, you know it very well from the community. Therefore the award happened due to a significant contribution of FreeSwitch to IP Telephony industry as well. As another user of FreeSwitch project, I take the opportunity to thank again to the people behind the project. This time, special wishes to Brian West - fast and smooth recovery, come back quickly to full "ultra wide band" capacity! If anyone is looking for more insights (for news, blogs, personal curiosity) about the event, just drop me an email! Cheers, Daniel -- Daniel-Constantin Mierla Co-Founder Kamailio SIP Server - http://www.kamailio.org Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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