February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

RTP jitter basic tutorial
by Madovsky 02 Feb '11

02 Feb '11

interesting http://toncar.cz/Tutorials/VoIP/VoIP_Basics_Jitter.html

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keeps receiving 407
by jesse 02 Feb '11

02 Feb '11

I set up a sipp uac client. here is the call flow messages: http://freeswitch.pastebin.com/Sx5Z4egM why does FS send 407 after 200 OK? thanks!

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voicemail.tpl and sip address
by Madovsky 02 Feb '11

02 Feb '11

which var can I use in voicemail.tpl to write correctly a sip address ? actually vocemail_caller_id_number shows only th username and not the domain Thanks

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Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true.
by Marcin Wojtowicz 02 Feb '11

02 Feb '11

I've found a solution file that creates the pcre_chartables.c source and compiled freeswitch from the latest git. The problem still remains, and it looks like it was brought up by somebody else here: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/063891.… Here are the build errors, in order of appearance: ------ Build started: Project: libpcre, Configuration: Release Win32 ------ pcre_chartables.c c1 : fatal error C1083: Cannot open source file: 'pcre_chartables.c': No such file or directory (other files here compile ok) ------ Build started: Project: FreeSwitchCoreLib, Configuration: Release Win32 ------ Generating switch_version.h (stuff here goes ok) Generating Code... LINK : fatal error LNK1181: cannot open input file 'E:\downloads\freeswitch-1.0.7\libs\win32\pcre\Win32\Release\libpcre.lib' ------ Build started: Project: mod_spidermonkey, Configuration: Release Win32 ------ mod_spidermonkey.c LINK : fatal error LNK1181: cannot open input file 'E:\downloads\freeswitch-1.0.7\Win32\Release\FreeSwitchCore.lib' FreeSwitchCore link error repeats many times for other projects, too many to list. It doesn't build. ------ Build started: Project: mod_managed, Configuration: Release_CLR Win32 ------ freeswitch_managed.cpp freeswitch_managed.cpp : fatal error C1192: #using failed on 'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the path specified.' freeswitch_wrap.2010.cxx freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the path specified.' mod_managed.cpp mod_managed.cpp : fatal error C1192: #using failed on 'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll' 'The system cannot find the path specified.' Generating Code... The code I downloaded was from latest.freeswitch.org, the tar.bz2 zip file. >From the dependencies it looks like FreeSwitchCore depends on libpcre, so all this cascades from the missing pcre_chartables.c file. Can you send it to me?

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dingalingissue
by budi wibowo 02 Feb '11

02 Feb '11

hi, anybody has "no voice" experience when call using mod dingaling? i try from my gmail account and call still normal but from FS always no voice. it was working perfectly before thx budi

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Problem with missing / duplicated DTMF
by Matt Stockton 02 Feb '11

02 Feb '11

I have having trouble with both missing and duplicated DTMF in Freeswitch. Here are the steps of how I am using it: 1. Leg A - I am calling out from my Freeswitch instance (through iCall), and I am calling an iCall number that is also connected to the same Freeswitch instance. 2. Leg B - The above call is routed through iCall and then answered by the same Freeswitch instance. 3. On Leg B, I play a file and attempt to get DTMF in a lua script. Here is the code of interest: callPin = session:playAndGetDigits(1, 10, 4, 30000, "#","/tmp/cw_17.wav", "", "\\d+"); 4. On Leg A, I send DTMF information in a lua script. Here is the code of interest. I initiate a delay between each digit: local newPin = ""; for i = 1, string.len(pin) do newPin = newPin .. string.sub(pin, i, i) .. "W"; end session:execute("send_dtmf", newPin .. "#@200"); ** Note that there is a session:sleep on Leg A before I send the DTMF to make sure i don't send it too early ** The problem is that the recognized DTMF on Leg B is wrong about 30% of the time. For example, if Leg A enters: 22063083, Leg B will get the DTMF digits 222063083. This is an example of duplication, but I have also experienced missing DTMF codes (and an occasional wrong code completely) I have messed with a bunch of DTMF settings in hopes of fixing this issue, but I cannot seem to find something that is reliable 100% of the time. _____________________________ Here are the DTMF settings I have looked at / messed with. I've tried various values for the dtmf-duration in the config (and in the send_dtmf command above)  <param name="dtmf-duration" value="500"/> <param name="dtmf-type" value="rfc2833"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="inbound-codec-negotiation" value="greedy"/>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU"/> ________________________ I have run fs_cli with event logging and the DTMF events that Freeswitch gets do correlate to the wrong value (e.g. the duplication / missing digits is noticable in the Freeswitch events as well). Also, I am not running any dtmf-related applications on the session before I give control to the lua scripts (e.g. not running start_dtmf) Has anyone experienced this type of issue? Or know what I can do to resolve it? My next step was going to be trying this against another provider besides iCall, but I figured I would see if anyone has encountered a similar problem before. Any help is appreciated. Thanks

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Modify Registration String to ITSP SipStation
by Brad Mina 01 Feb '11

01 Feb '11

I'm having trouble registering a SipStation account. They require username:password@trunk1.freepbx.com<username%3Apassword(a)trunk1.freepbx.com>to be sent to the registrar. I've done a bit of searching but was unable to find any specific details on how to do this. My config is as follows: <include> <gateway name="SipStation"> <param name="username" value="myUsername"/> <param name="password" value="somePassword"/> <param name="realm" value="trunk1.freepbx.com"/> <param name="from-user" value="TriTelComm"/> <param name="proxy" value="trunk1.freepbx.com"/> <param name="expire-seconds" value="800"/> <param name="register" value="true"/> <param name="retry-seconds" value="60"/> <param name="context" value="public"/> </gateway> </include>

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H323 Setup
by Tim King 01 Feb '11

01 Feb '11

> > I am following http://wiki.freeswitch.org/wiki/Mod_h323 and trying to get > H.323 setup. I installed using buildopal.sh script in the build directory.I > am not finding h323.conf.xml anywhere except the sample in > /usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/h323.conf.xml. Do > I need to create this file and if so where should it be? > Thanks Tim

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Load mod_xml_rpc via cli?
by Addison Martin 01 Feb '11

01 Feb '11

I seem to be able to load mod_xml_rpc at runtime, but netstat -a isn't showing port 8080 as listening. Is it possible to do this at runtime, or is a restart of the switch required? I'm buidling a click2dial greasemonkey script, and the xml_rpc api seems the way to do it without a bunch of client side requirements. Regards, Nik

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freeswitch.com is returned to us!
by Anthony Minessale 01 Feb '11

01 Feb '11

http://www.freeswitch.com/ Thank you everyone who helped with this! -- 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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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