March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

sip-force-contact: which to choose?
by Fabio Pietrosanti (naif) 22 Mar '10

22 Mar '10

Hi all, i read in latest source code of FS (of yesterday) that for the handling of sip-force-contact now there is a new option: NDLB-connectile-dysfunction-2.0 Which are the improvement of NDLB-connectile-dysfunction-2.0 respect to NDLB-connectile-dysfunction? I read that the 2.0 also make the users registered with "AUTO-NAT 2.0". Fabio

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Now Hiring for CudaTEL UI Programmer
by Anthony Minessale 22 Mar '10

22 Mar '10

As most people know, the CudaTEL uses FreeSWITCH internally as the telephony engine. We're now looking to expand the development team: We're looking for a UI programmer with jquery experience. See: http://jobs.jsninja.com/job/1598/jqueryjs-ui-developer-at-barracuda-network… Anyone interested, please apply to this forum or email jobs(a)freeswitch.org Thanks. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:+19193869900

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Re: [Freeswitch-users] Open source softphone that can handle g729
by Emil Ivov 22 Mar '10

22 Mar '10

> sip communicator is crap. Its trying to do way too much and does not > a single thing very well at all. Well, that's a bit vague. Maybe if you've had/have specific issues with it, you could drop us a note so that we could fix them. In any case the project has improved a lot during the last years, and even months so you might want to give it another try one of these days. > If they have G729 its not legal to use. Yes we do have it, and it was implemented as a part of our participation in GSoC 2009. People can use it perfectly legally as long as they buy the corresponding licenses (just as with any other client). Currently use of the codec is disabled by default and whoever needs it has to enable it in the code, after acquiring the necessary license. Cheers, Emil > > /b > > On Mar 20, 2010, at 8:24 AM, Raffaele P. Guidi wrote: > >> I believe sip communicator does handle g729. It has a strong >> community and a plugin model (based on osgi) that makes it easily >> (ok, kinda...) customizable. Of course the >> FSComm/givesomemoneytotheproject option is the best one. In any >> case: http://sip-communicator.org/ >> >> Ciao, R >

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Migrating voicemail to new freeswitch instance
by Adam Ford 22 Mar '10

22 Mar '10

How would I migrate people's custom recorded voicemail greetings to a new install of FreeSWITCH? I finally got upgraded from 1.0.4 to SVN trunk. Thanks, -Adam

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How to upgrade configuration files when upgrading FreeSwitch
by Yehavi Bourvine 22 Mar '10

22 Mar '10

Hello, Once a week I upgrade our production system by retreiving freeswitch-1.0.5-latest.tar.gz, building and installing it. This updates all the binaries, but I am left with the current config files (which is good, as some of them contains locally edited data). The problem is how to detect (and merge) changes in the configuration files, like new parameters. I can do manual diffs, but it is a time consuming task. What is the BCP of doing such an upgrade? Thanks! __Yehavi:

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Way to *always force* media relay across FreeSWITCH
by Fabio Pietrosanti (naif) 22 Mar '10

22 Mar '10

Hi all, i am using FreeSWITCH with mobile phones and have problems in FS sometimes acting as a media relay, sometimes not. The mobile phones are behind operator's 3G/2G data networks that are "natted" . The FS just need to act as a SIP server handling: - Signaling (with SIP/TLS) - Media relay NO transcoding is done. (disabled) Mobile devices use an obfuscated RTP protocol (to bypass VoIP blocking by mobile operator) and in order to let FS relay it we use: inbound-proxy-media=true . That way FS does not even check the RTP header but just relay udp packets on agreed ports. I am having a strong problem, sometimes devices get one-way audio and doing debugging we found out that sometimes FS rewrite correctly the SDP header but sometimes it does not. We are using: <param name="apply-nat-acl" value="nat.auto"/> <param name="aggressive-nat-detection" value="true"/> I would like FS to act like Asterisk canreinvite=no that force the media to go trough it always rewriting SDP headers and to make the RTP flow goes across Asterisk. I also tried to add this in order to force the rewriting: <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> However given 2 peers sometime i see the status of the peer (sofia status profile internal) shown as "Registered(AUTO-NAT)" and sometimes as "Registered(TLS)" . Some times the traffic between the peers is not correctly relayed and there is an asymmetric one-way audio situation: peer-A correctly send RTP traffic to FS that relay it to peer-B peer-B send RTP traffic directly to peer-B (because FS did not rewrote the SDP header related to the target IP of RTP). From our basic diagnosis it seems that FS sometimes detect devices as "behind NAT" but sometimes it does not. Is there a way to certainly and precisely force all connections to be relayed across the FS like we was doing with "canreinvite=yes" of asterisk? I just would like a static set of settings without the dynamic methods of NAT detection. Just everything (RTP relay) going trough FS. Is that possible? Fabio Pietrosanti

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Radius Authentication/Authorization
by Abid Saleem 22 Mar '10

22 Mar '10

Hi, I am trying to find out if FreeSwitch supports Radius Authentication/Authorization with free radius or any other radius server. If Yes, How can I configure and If No, what are any alternative workarounds to make it work for prepaid call authentication/authorization. THanks in advance for help. Regards -----------Abid SaleemSr. Product Manager _________________________________________________________________ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969

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Newbie Questions
by Ken Fulmer 22 Mar '10

22 Mar '10

1. We would like to transcode from g711mu internally to g729 for PSTN calls. Is this possible? 2. We are using sipX as an open source PBX. We'd like to use FreeSwitch purely as a B2BUA for connectivity to the PSTN. Our Polycom phones use SIP REFER messages to do call transfers and our ITSPs can't handle those. So, we've tried the "under the hood" FreeSwitch within sipX. It works for many applications but we need to scale a unified system for larger customer deployments. If we want to create a trunk to an ITSP and a trunk to an internal PBX, would we use the external gateway XML files for both? We don't plan to point any internal phones to the FreeSwitch box (we're using sipX as an internal PBX). 3. If we use multiple ITSPs, can we assign a priority to each one, so one will be chosen first? 4. I understand no performance metrics are posted. Can anyone share their experiences with passing RTP through the server? Does RTP affect performance in a big way (very open ended question, I know). We are looking for ball park answers so we'll have some idea of scalability for our designs. Thanks, Ken Fulmer

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Unauthenticated users vs activated users
by Fabio Pietrosanti (naif) 22 Mar '10

22 Mar '10

Hi all, on my system i completely disabled any kind of users authentication with passwords. However i still need to check if a specific URI is "enabled" to login (even if unauthenticated). So i still need to check if the user can REGISTER or not, checking if it's "username" is enabled to login or not. However when i disable authentication FS does not anymore make xml_curl lookup, reasonably, because there's no authentication to be done. However i need to check in a database if the user (checking only the login-name) is authorized to login with user/password. Is there some workaround and/or particular configuration with which this kind of logic could be implemented/enforced or does it require code modifications? If it required introducing a feature to FS i am available to provide financial support for consulting to implement such feature. Waiting for yours Fabio Pietrosanti

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mod_dingaling endless_playback issue
by Jingwei Yang 22 Mar '10

22 Mar '10

Hello, I just upgraded to the latest svn (r17063) today and found an issue of mod_dingaling. I use the command below to originate a call to a gtalk client and make him hear a piece of endless music: originate dingaling/gmail.com/jingwei.yang@gmail.com 33333 Here's how the dialplan is defined. <action application="sleep" data="2000"/> <action application="info"/> <action application="endless_playback" data="customized/to_client_hold_call.mp3"/> After the call is answered, the client can't hear the music at all and after 30 seconds or so, the call is dropped automatically. I checked this dialplan with a skypopen client and it worked with no prob (and it used to work in r16600). Please let me know where goes wrong. Thanks and best regards, -Jingwei

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