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Call doesn't work while registration work for a VOIP provider
by Albano Daniele Salvatore - Lavoro 20 Nov '09

20 Nov '09

Hi, i'm trying to configure freeswitch with a VOIP provider, exsorsa, that uses OpenSER. Exsorsa use as own gateway, another provider, Eutelia, that it uses Cisco (or, at least, this appears in headers). Short story: ------------ If i try to setup my Eutelia account all works perfectly while if i try to setup Exsorsa account registration works fine while calling not: when fs send the ACK, as answer to a OK (sip code 200), that is sended from exsorsa as answer to an INVITE, exsorsa send back a BYE. Long story: ----------- I put call log on pastebin with debug and sip_trace enabled for external sip_profile and with log level on debug on fs console. Registration log, here all is ok (or at least it seems to be ok) http://pastebin.freeswitch.org/11176 Annoyng message that comes up every 30 seconds http://pastebin.freeswitch.org/11177 Call log http://pastebin.freeswitch.org/11178 As you can see from call log all works fine until fs send back the acknowledgment message (line 451 on last log). Can this depend on the annoyng message that comes up every 30 seconds? Here my external sip profile config http://pastebin.freeswitch.org/11180 while here exsorsa gateway config http://pastebin.freeswitch.org/11181 Any helps is really appreciated! I'm fought with it all the day!!! Best Regards, Daniele

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TFTP Server & Cisco 7540
by Dave Stevenson 20 Nov '09

20 Nov '09

Hi, I have just about got FreeSwitch working with a Cisco 7940 Phone. After much reading, I worked out that I needed a TFTP server on the network that would supply the phone with it's SIP personality and config etc. I have been able to get the phone working and realise that the TFTP server needs to be available every time the phone loses power etc. At the moment, I have the TFTP server running on a temporary machine but it would be neater if it ran on the same machine as FreeSwitch. This will be a very small FreeSwitch installation, so, ....... Is there any reason why I should not try to run FreeSwitch and the SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't think the server should put much load on the machine but wondered if there were any other reasons why this is a bad idea ? In addition, while I have the phone working - I get a status message on boot ... "W310 2 Errors(s) Parsing SIPDefault.cnf Can anyone tell me how to locate the errors in this file please ? (I have posted it to the Pastebin) Regards Dave

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mod_bv16/32 removed. Added mod_bv
by Brian West 20 Nov '09

20 Nov '09

We have removed the two modules using the reference code from BroadVoice and added a lib with a new interface from Steve Underwood and mod_bv.c using this lib... We know their is ONE last bug to be fixed in the lib before its working so please do not open any jira's if you try to run it because it will crash right now. Thanks for your understanding and once this is fixed it'll work with aastra and x-lite on both 32bit and 64bit systems without any issues. Thanks, Brian West

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Another Group Question - on VoiceMail
by Dave Stevenson 20 Nov '09

20 Nov '09

Hi again ! I have FreeSwitch configured such that if someone dials in from the PSTN line, a group of phones ring. If nobody answers, the group extension number (100) picks up the call and voice mail kicks in. So far, so good, each of the individual phones logs a missed call and anyone in the group can call into voice mail and go to the extension 100 mailbox to check if there are any messages but the individual phones are not notified that a Voice message is waiting. Is there any way that each extension in the group can be notified that a group Voice Mail is waiting to be picked up so that each phone shows the message waiting indication ? Regards Dave

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upgrading to latest SVN
by Ujjval Karihaloo 20 Nov '09

20 Nov '09

Getting error below..not sure whats wrong..which line number in what file does this refer to? [root@ss_freeswitch log]# freeswitch 2009-11-19 20:15:44.725118 [INFO] switch_event.c:568 Activate Eventing Engine. 2009-11-19 20:15:44.727095 [DEBUG] switch_event.c:556 Create event dispatch thread 0 Cannot Initialize [[error near line 733]: missing >]

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Setting up Conference with Moderator
by Ujjval Karihaloo 20 Nov '09

20 Nov '09

Hi I have the Basic Conferencing working. Here is my Dial Plan. I want to be able to setup a Moderator PIN different from other participants, when I add the moderator flag it logs me in directly w/o asking for a PIN.. action application="conference" data="conference.conf@wideband+flags{moderator}+159753<mailto:conference.conf@wideband+flags%7bmoderator%7d+159753>"/> DialPlan is below for the normal user, and it asks for the PIN with below settings. Ujj Inbound from SS - start --> <extension name="simplesignal">  <condition field="destination_number" expression="2142349127">  <action application="conference" data="conference.conf@wideband+159753<mailto:conference.conf@wideband+159753>"/> </condition> </extension> <!-- Ujj Inbound from SS - end And I am using the existing conference.conf.xml file in the auto_loads directory. Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [cid:image001.jpg@01CA535A.E43E9870]<http://www.simplesignal.com/>

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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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