October 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Early media beginning clipped for certain callers
by Mario 19 Oct '10

19 Oct '10

I have replaced what the incoming caller hears (ringing) with a greeting, instructions, ringing. Works great but for some callers, about 1 second is clipped, it seems to do with who is calling: Caller A cell phone & B land line works every time and is never clipped. Caller C cell phone & D land line clips every time. I added up to a 5 second "sleep" before the media to test but it had no effect telling me the other end is clipping the audio when it starts. Any ideas on what I might try? I can try to lengthen the audio but I didn't want to make the working ones listen to extra silence. Thanks! Mario

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Re: [Freeswitch-users] Compile on Windows
by Tim B 19 Oct '10

19 Oct '10

Ok and update... I was able to try to compile on an WinXP x86 from GIT using VS2008. But the following failed.... - Win7 x64, GIT, VS2010 - Win Server 2008 x86, GIT, VS2008 and VC++ Express 2010

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Remove rport from Via header
by Nicolas Brenner 19 Oct '10

19 Oct '10

Hello, I need to either remove the rport parameter from the Via header on an initial invite or set its value to a specific port. I've looked through the mailing archives and the wiki but I haven't found anything. I appreciate any help or pointers I can get, thanks. Nico

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Gateway responding to random port (rport problem?)
by Nicolas Brenner 19 Oct '10

19 Oct '10

Hello everyone, I'm having an issue with a VoIP provider where the gateway will respond to any random port instead of 5080. I sent the corresponding trace to the provider and they told me to "disable rport and nat". I removed the auto-nat parameters in the corresponding sip profile, but I keep having the same issue (I don't think it has anything to do with nat anyway). On the trace I noticed that INVITEs sent by FS have an empty rport parameter on the Via header, but according to what I've read, this is standard and it indicates that the response should be sent to the originating port (5080), which is not being done by the gateway ("Avaya SIP Enablement Services"). How may I go about solving this? should I set the rport parameter to something specific (eg. 5080)? or remove it altogether? Thanks, Nico

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Are there any rpath based FS distribution?
by Fabio Pietrosanti (naif) 19 Oct '10

19 Oct '10

Hi all, does anyone know if there is some rpath (www.rpath.com) based FS distribution that can be easily installed as an appliance by using a cdrom? Fabio

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FreeSwitch as SBC, no registration?
by joedotslash 19 Oct '10

19 Oct '10

Hello all, I'm looking to implement the following scenario: Step 1 SIP Server A sends INVITE to port 5060 over IPv6 to FreeSwitch: 2001::1 --udp/tcp--> 2001::2:5060 Step 2 FreeSwitch SIP NATs the INVITE and sends it out IPv4 to SIP Server B on port 6000 1.1.1.1 --udp/tcp--> 1.1.1.2:6000 Step 3 All SIP Server B responses within this SIP dialog are sent back to SIP Server A along the same path. REGISTER messages WILL NOT be processed by FreeSwitch. They are handle elsewhere. There is more to this but I want to accomplish this first call flow go from there. Is this possible with FreeSwitch? Can someone point me toward a resource on how to get this created? The following looks like a good start but it seems it's setup for users to register. I would prefer not to have this. http://wiki.freeswitch.org/wiki/SBC_Setup Thanks! Joe

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Re: [Freeswitch-users] Compile on Windows
by Tim B 19 Oct '10

19 Oct '10

Thanks guys... I have tried building... Nightly Snapshot of Git Source freeswitch-snapshot.tar.gz the official release http://files.freeswitch.org/freeswitch-1.0.6.tar.gz and git clone with autocrlf=false still getting errors and failed builds.... ========== Rebuild All: 58 succeeded, 76 failed, 18 skipped ========== Now is there an specifics I need to set in VS2010 for my c++ options, etc? I am a c# developer so I don't do much with c++, so I never messed with any of the options there. I did download the windows 7 sdk so I would have all the windows header files. Tim

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Limit [ERROR] question
by DJB International 19 Oct '10

19 Oct '10

I have noticed once in a while in the log file with the following error: [ERR] switch_limit.c:86 Unset limit backendlist! What exactly would cause this error? Thank you, Dorn B.

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Sofia gateway incoming problem
by Maxim Balabaev 19 Oct '10

19 Oct '10

I`m having really strange problem. I connect new provider, changed all gateway settings but i can`t take calls. While debuggins external and so as global no invite packet is showing up. I even tryed tcpdump - absolutely nothing. Just pings. Outcoming are ok, register is ok. If I switch to asterisk - no problem occures. Support says that server refuses connection, while another gateway works perfectly. I don`t know what to do... May be this can be usefull to. Prov have multiple realms on server and accepts only @myprov.com format. While to asterisk invite is thrown with sipid@myip. I tryed changing server hostname to ip, but still nothing. Here is my conf: http://pastebin.com/qKqAv38a <http://pastebin.com/mMrj33pN> Really appritiate any help.

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Conference leg recording
by Scott Brown 19 Oct '10

19 Oct '10

I'm wondering if there currently is a way to record one leg of a conference call. I have tried several different methods, but always seem to get the entire conference. Thanks! Scott

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