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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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freeswitch and android
by Brian May 22 Mar '10

22 Mar '10

Hello, Anybody here got freeswitch working with Android? If so, what app did you use? I tried sipdroid which looked very good, but unfortunately it seems it answers incoming phone calls immediately and fakes a ring tone. I filled a bug report, but it was closed as invalid, so far without explanation: http://code.google.com/p/sipdroid/issues/detail?id=383 Thanks -- Brian May <brian(a)microcomaustralia.com.au>

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Re: [Freeswitch-users] rtp timeout yet
by Madovsky 22 Mar '10

22 Mar '10

For example, I set rtp-start and end port in switch.conf.xml from 16000-32000 but When I register a phone it uses port above 32000. example on a register sip trace : Via: SIP/2.0/UDP 10.10.10.10:52641;rport=52641 Any idea ? ----- Original Message ----- From: "Madovsky" <infos(a)madovsky.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Sunday, March 21, 2010 11:12 PM Subject: Re: [Freeswitch-users] rtp timeout yet > Hi Metik, > > Thanks for your answer. Just wondering which rtp port FS use really. > on the sip trace I can see often rtp port up to 32768 (like 50434 etc..) > so it should be a problem since I use apache with port 50050:60100 on > the same IP as FS. > > > ----- Original Message ----- > From: "Metik" <freeswitch-users-list(a)metik.com> > To: <freeswitch-users(a)lists.freeswitch.org> > Sent: Sunday, March 21, 2010 3:46 PM > Subject: Re: [Freeswitch-users] rtp timeout yet > > >>I haven't followed this particular thread very closely but are you sure >>there are no port conflicts exhibited in the SDP? This is UDP after all >>and unlike TCP, binding is exclusive and port selection would then need to >>take in account those ports already in use (the endpoint and FS would be >>pretty much flying blind if the endpoint is not a local process). >> >> I would suggest reviewing your endpoint configurations and making sure >> that they are using unique (rtp/media) port ranges. One would think that >> if the endpoint shares the same global IP address as used by FS and it is >> not a (softphone) process running on the same physical box, then it would >> be communicating with FS using its private IP. Does both the signaling >> and media traffic traverse (i.e. routed) through a broadband router that >> sits in front of both FS and the endpoint or is it sent (i.e. forwarded) >> directly? >> >> -metik >> >> Madovsky wrote: >>> last trunk, >>> remove current conf folder and reinstalled the default one after make >>> install. >>> I stopped iptables, connect 1000 to 1001 on port 5060, but >>> after rtp-timeout it closes yet. >>> sip trace after the rtp timeout >>> SIP/2.0 481 Call Does Not Exist >>> I start to be very frustrated.... >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org

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rtp timeout yet
by Madovsky 22 Mar '10

22 Mar '10

last trunk, remove current conf folder and reinstalled the default one after make install. I stopped iptables, connect 1000 to 1001 on port 5060, but after rtp-timeout it closes yet. sip trace after the rtp timeout SIP/2.0 481 Call Does Not Exist I start to be very frustrated....

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Re: [Freeswitch-users] rtp timeout yet
by Madovsky 22 Mar '10

22 Mar '10

Apparently there s a timeout everytime I register a phone on a public IP. for phone behind a firewall (outside FS network) it works well. ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, March 21, 2010 3:09 PM Subject: rtp timeout yet last trunk, remove current conf folder and reinstalled the default one after make install. I stopped iptables, connect 1000 to 1001 on port 5060, but after rtp-timeout it closes yet. sip trace after the rtp timeout SIP/2.0 481 Call Does Not Exist I start to be very frustrated.... -------------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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