November 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Radius for registration
by Lon Baker 19 Nov '09

19 Nov '09

Hi everyone, I want to verify what the wiki says, you can use a radius server as the data source for your registrations? Lon

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Call latency in conferences and echo test increases over time
by Robert L Mathews 19 Nov '09

19 Nov '09

I'm using FreeSWITCH 1.0.4. When I make a call from a SIP phone to either a conference or an echo test on the FreeSWITCH server, the latency ("lag") starts off very low -- a fraction of a second. But as several minutes of time goes by, the lag increases. After, say, 15 minutes, the lag will reach a couple of seconds, making conference calls unusable. This does not happen on pure SIP-to-SIP calls, even when FreeSWITCH is handling the RTP media. If I hang up and immediately call back in (even to the same conference), the lag is reset to almost zero. If I put the call on "hold" and take it off hold, the lag is also gone. During testing, I've found that this may be related to the freeswitch app on the server not getting all the CPU time it wants. If I suspend the freeswitch process for two seconds and then resume it, the sound stops for two seconds, as I'd expect. But the echo/conference calls that were active are then lagged by two seconds until they hang up (or get put on hold), even after freeswitch is resumed and getting all the CPU time it needs. This is easily reproduced by making a SIP call to the echo test module, then: pkill -STOP freeswitch; sleep 2; pkill -CONT freeswitch Any echo test or conference call that was in progress will then be permanently lagged by two seconds. However, any SIP-to-SIP calls that were in progress will not become lagged. Of course, killing it with -STOP is an artificially nasty thing to do. But it effectively just prevents it from being scheduled on the CPU for a short period of time, and I can duplicate the same behavior (more gradually) by just increasing the load on the machine to the point that the freeswitch app isn't getting much CPU time. Just for the record, I get the same results from several different phones and several different Internet connections, all of which have a ping latency of under 40 ms to the server. This problem does not happen using the same phones and network connections to an asterisk server. Throwing out an even more complicated example that I've encountered: If I have a SIP-to-SIP call going from party A to party B and I stop the process for two seconds, it doesn't permanently introduce lag to that call, as I mentioned. But if a third person (party C) starts eavesdropping on the call and presses "3" to make it a three way call, and then I suspend it for two seconds, the call between A and B isn't lagged, but what party C hears and sends *is* lagged. Any ideas on how to fix this? Do other people see the same thing happening? As I said, the gradual increase in lag over a long period of time makes long conferences unusable, unfortunately. -- Rob

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Need help configuring our FreeSWITCH instance
by John Platts 19 Nov '09

19 Nov '09

I have installed FreeSWITCH on our server, and need some help configuring our FreeSWITCH instance. All of the numbers associated with our FreeSWITCH instance are in the format: 1NPANXXXXXX (where NPA is the area code, and NXXXXXX are the last 7 digits of the phone number). I need the following configuration: Calls coming from our IP to IP gateway into our FreeSWITCH instance needs to be routed to the endpoint that is registered with FreeSWITCHCalls coming from any of the registered SIP endpoints need to be sent to the appropriate destination. The appropriate destination for any number that is not registered with FreeSWITCH is our IP to IP gateway.Our IP to IP gateway does not require any SIP registration or authentication.G.729 (but not G.729 Annex B), G.711 mu-law, and G.711 A-law need to be enabledSIP registrar enabled for registering endpoints other than our IP-IP gatewaySIP traffic needs to be accepted to and from both the IP-IP gateway and from the registered SIP endpoints. How do I get the above configured in FreeSWITCH? _________________________________________________________________ Windows 7: I wanted simpler, now it's simpler. I'm a rock star. http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24…

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Extension Configuration - XML File Entries for Group configuration
by Dave Stevenson 19 Nov '09

19 Nov '09

Hi, Can someone please help me understand a little more about Group configuration ? I believe that Group Membership is configured in the \conf\directory\default.xml file I've done this and the caller groups seem to work fine. However, each extension in the \conf\directory\default directory, e.g., 111.xml also has an entry for "callgroup" Can someone explain what the difference in these two options is please ? regards Dave

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Re: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP
by Jerry Richards 19 Nov '09

19 Nov '09

Hello, I just pasted a log in the Pastebin with Freeswitch logging enabled. Does anyone know a way to prevent FS from connecting the call prior to the callee answering? Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Thursday, November 05, 2009 3:50 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: Want 183 w/SDP, but Get 200 w/SDP I am trying to make a call through a Gateway that sends the INVITE with no SDP and ONLY wants the 200 OK w/SDP when the callee answers. For some reason, Freeswitch answers the call with 200 OK w/SDP even before the callee answers the phone. Is this to provide ringback? Can I disable that action? Best Regards, Jerry

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mod dptools record problem - hangup channel with invalid file path
by William Kendi ... 19 Nov '09

19 Nov '09

While I was testing the "mod dptools record" application using invalid file paths, i noted that the "mod dptools record" application terminated the call. I am currently looking for a way to change this behaviour. Any suggestions? Can this be done?

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How to implement mod_lcr + mod_limit
by Ahmed Munir 19 Nov '09

19 Nov '09

Hi, I've worked on setup for carriers routing using mod_lcr + mod_nibble + mod_xml_curl and mod_xml_cdr. The setup is working fine as I desired. Now I want to include mod_limit in to my setup. As I read the wiki pages of mod_limit I want to know how can I limit the calls per destination basis while running mod_lcr? Because LCR is routing to different carriers, how can I call mod_limit in mod_lcr? Kindly advise this issue soon. -- Regards, Ahmed Munir

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store registration info in memcache
by Woody Dickson 19 Nov '09

19 Nov '09

Hi, Is there anyway to store registration info in memcache instead of sqlite? By doing that, it is possible for multiple freeswitch to share the same user registration info. Is there anyway I can intercept the registration success/failure event and write stuff to memcache myself? thanks, woody

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Question about odbc support
by Helmut Kuper 19 Nov '09

19 Nov '09

Hi, does anybody know how to check the affected rows caused by delete, insert or update sql statements in FS? To do this with sqlite3 there is a function called switch_core_db_changes(). regards helmut

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Changing User-Agent String
by Ujjval Karihaloo 19 Nov '09

19 Nov '09

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#User_Agent_.5Buse… As per the above link, we can change the User Agent String, but I added this param name but does not seem to work. [user@freeswitch autoload_configs]$ vi sofia.conf.xml <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/>  <param name="debug-presence" value="0"/> <param name="user-agent-string" value="Test Server"/> </global_settings>  <profiles> <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml"/> </profiles> </configuration>

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