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Re: [Freeswitch-users] FS Docs Questions - wiki searching
by C. Mylonas (Mr VoIP) 23 Jun '08

23 Jun '08

http://dev.mysql.com/doc/refman/5.0/en/fulltext-fine-tuning.html It could be something to do with the minimum length of a word when doing a full text search. Kind Regards, Chris Mylonas

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Outbound calls to external domain
by Sandeep Ammaladinna 23 Jun '08

23 Jun '08

Hi all, Is it a mandatory requirement that for all outbound calls to an external domain (domains that freeswitch is not responsible for), an entry in the dial plan is required. To be more precise, is it a pre condition that the name of the external domain need to be know before making a call. If YES, then I feel it is a bad design, because each time a user wants to call a user from different external domain a new entry in dial plan is required. If the answer is NO, please tell me what is the configuration required apart from DNS records, to be there at the freeswitch. Thanks and Regards, -Sandeep A

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FS Docs Questions - wiki searching
by Michael Collins 23 Jun '08

23 Jun '08

Guys, After Anton's post to the users list I did a search on "tdm" and "TDM" on the wiki and got no results, not even page text results. This is bad because the second sentence on the OpenZAP wiki page is: The project implements a TDM abstraction library to provide an interface to TDM hardware as well as various signaling for T1/E1 interfaces. Furthermore, I just noticed that when you do a wiki search, whether it returns results or not, the user is presented with an option to go to this page: http://wiki.freeswitch.org/index.php?title=FreeSWITCH_Wiki:Searching&act ion=edit This page does not exist. :-( Question #1: is there some sort of indexing process that we need to do so that page text searches will find stuff like "TDM"? Question #2: any thoughts on what should be in the "FreeSWITCH_Wiki:Searching" wiki page? Thanks, MC P.S. I've added a new page for this Wednesday's weekly call: http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_25

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H323 support chip in
by Brian West 23 Jun '08

23 Jun '08

http://www.chipin.com/contribute/id/c648c4411bfb7861 Here is the URL to chip in for the H323 support in FreeSWITCH. I'm a bit disappointed because so many of you out there were asking, begging and really pleading for solid h323 support. Please chip-in and lets make it happen. Thanks, Brian PS don't forget cluecon.

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FS benchmark test
by Santiago Gimenez Ocano 23 Jun '08

23 Jun '08

We have been doing tests in order to establish benchmarks. This is the scenario: SIPp -------> FreeSWITCH ----- > Destination SIPp: SIPp is using the uac scenario with a delay of 15000ms. INVITE ----------> 100 <---------- 180 <---------- 183 <---------- 200 <---------- E-RTD1 ACK ----------> Pause [ 15.0s] BYE ----------> 200 <---------- FreeSWITCH: We have used two different servers: FS1: Virtualized Processor: Xeon @ 3.00GHz OS: Debian Kernel version: 2.6.18-6-686 FS version: 1.0 Preemption: disabled FS2: Not virtualized Processor: Intel Dual Core @ 3GHz OS: Fedora Linux 9 Kernel version: 2.6 FS version: 1.0 Preemption: enabled Both servers are running in no console mode and with ulimits, as http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations and http://wiki.freeswitch.org/wiki/Long_term_testing_chevymanjosh mention. Also, both servers are using the same dialplan which access to a database via Java Script. This database receives a query with the destination number and respond with a list of destination IP. Then the calls is bridged. The results: In FS1 we have had a load average of 4.00, retransmission and errors with 7 attempts per second and 100 sessions aprox. In FS2 we have had a load average of 4.00, retransmission and errors with 13 attempts per second and 200 sessions aprox.. Conclusion: We are worried about this behavior and think about this figures are quite low in comparison to other experiences that we have seen in Wiki (1000+ calls and 50 attemps/sec) or articles. JS is not the problem, we are getting the same results using a simple dialplan. <include> <context name="default"> <extension name="init" continue="true"> <condition field="destination_number" expression="^(.*)"> <action application="set" data="accountcode2=8888"/> <action application="set" data="contexto=dp3"/> <action application="set" data="called=15555555555"/> <action application="set" data="export_vars=uuid,destination_number,accountcode2,contexto,cal <action application="bridge" data="sofia/default/1555555555555(a)192.168.1.182"/> </condition> </extension> </context> </include> So, all is pointing in some bad configuration that we are missing. Please, if you have any idea it would be appreciated. Thanks in advance Santiago _________________________________________________________________ Do more with your photos with Windows Live Photo Gallery. http://www.windowslive.com/share.html?ocid=TXT_TAGLM_Wave2_photos_022008

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Regular expressions
by Sandeep Ammaladinna 23 Jun '08

23 Jun '08

Hi, Can anybody help me out with a regular expression (in the context of writing a dial plan) to extract the host part of a SIP URI. Suppose I have a sip uri like "sip:alice@example.com" or alice(a)10.0.16.20 what is the procedure to extract the host part? I know the procedure to extract the user part, ie, "$1@". But what is the procedure to extract the host part???????? Please help me, I am stuck. Thanks and Regards, -Sandeep A

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Calls to external domains (outside home domain)
by Sandeep Ammaladinna 23 Jun '08

23 Jun '08

Hi all, I am very new to Freeswitch and facing couple of issues related to making outbound calls to external domains. Below is a detailed explanation about the scenario I am trying and my setup. 1) I have freeswitch installed and running. I have configured a domain within freeswitch by the name "freeswitch.com". 2) I have registered couple of users' user1(a)freeswitch.com and user2(a)freeswitch.com . 3) Calls are going fine between user1 and user2. Now I have an external proxy. I am using "OPENSBC" as an external proxy. 1) OpenSBC is running and I have a domain within opensbc by the name "opensbc.com". 2) I have a user registered as user3(a)opensbc.com. Now the problem is, when I make a call from user1(a)freeswitch.com to user3(a)opensbc.com , the call is not getting placed. DNS server is running and corresponding NAPTR, SRV and A record are also configured. My assumption is if we want to make calls within "freeswitch.com" domain, we require dial plan to be configured for routing purpose. But if we are making calls outside "freeswitch.com" (basically calls to domains for which freeswitch is not responsible for) transparently DNS will be called and call has to establish. My assumption here is we need not configure any xml files at freeswitch. If I am wrong please tell me where I am wrong and what I need to do in order to establish a call between user1(a)freeswitch.com and user3(a)opensbc.com . Thanks and Regards, -Sandeep A

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donald k has added you as a friend on BluBet
by donald kingsley 23 Jun '08

23 Jun '08

I challenge you to BluBet! https://www.blubet.com/invitepath/StdA/784b289440f211dd945e02bf453b928c donald k --- This email was sent to you by BluBet user donald k (kickamyassa(a)gmail.com) This email is never sent unsolicited. If you believe you have received this in error, please contact us at: privacy(a)blubet.com If you do not want to receive future emails from BluBet, click below: http://www.blubet.com/optout.aspx?emailid=784b2895-40f2-11dd-945e-02bf453b9… BluBet, Inc., 208 Utah Street, Suite 404, San Francisco, CA 94103, USA.

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deploying freeswitch
by Pete Kay 23 Jun '08

23 Jun '08

Hi, I am hoping to get some deployment insights from people who have implement freeswitch on a large-scale. I have followed the wiki in setting up Xen and Ultramonkey to load balance two freeswitch. But since freeswitch is stateful and two freeswitch can't share info such as fifo, I am wondering if it is a good idea to do just Hot Standby using Linux-HA instead of load balancing. The problem with just Hot Standby is that I can't load balance traffic to multiple machines, which eventually limits the number of concurrent I can serve. Could someone gives suggestion as to how to best approach this problem? Thanks in advance for your help. Best Regards, Pete

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OPTIONS sip
by Ivan C Myrvold 22 Jun '08

22 Jun '08

My FreeSwitch box is sending these OPTIONS sip messages to my Cisco 7960 phone at 192.168.207.160. What does it mean? It is sending these messages even when I unplug the Cisco phone: http://pastebin.freeswitch.org/4669 Ivan

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