June 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

iSoftPhone strange behaviour
by Ivan C Myrvold 24 Jun '08

24 Jun '08

I have a softphone, iSoftPhone, I am trying out, but when registered to FreeSwitch, it behaves strangely. That is probably the softphone's fault, but I thought I would report it here, so I could provide the developer of this phone with some more intelligent response. I have iSoftPhone on the same subnet as FreeSwitch, and it is registered ok. I then call from iSoftPhone to another registered phone on the same subnet, and answers. The SIP messages are here: http://pastebin.freeswitch.org/4688 The call are up, and I have nromal audio. I see that iSoftPhone is sending regularly the message SIP/2.0 486 Busy Here to FreeSwitch. I have reported this to the developer, but never got any response back. After 1 minute, FreeSwitch sends a new INVITE sip:imyrvold@192.168.207.101:5060 SIP/2.0 to iSoftPhone, maybe because of the "486 Busy Here" message? I lose audio now, and I have to hangup the call. Can anyone comment on the above behaviour? Ivan

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iLBC question
by Alex Gusak 24 Jun '08

24 Jun '08

We have some problems with codec iLBC. There is a scheme for passing the call: +--------+ +------------+ ClientA----->| |-------->| | | SER | | FreeSWITCH | ClientB<-----| |<--------| | +--------+ +------------+ At FreeSWITCH allowed codecs iLBC@30i, PCMA. RTP runs through FreeSWITCH. When a clients set different codecs (eg A: iLBC, B: PCMA), or both customers PCMA codec installed both the client hear each other. If two clients installed codec iLBC -- clients do not hear each other, while RTP traffic FreeSWITCH passes through (in this case from one client comes payload type 97, and to another client FreeSWITCH sends payload type 102). If we include proxy_media = true, then clients hear each other (FreeSWITCH sends payload type, which came from the customer A). But then not working transcoding. Calls test at several softphones: zoiper, XLite, SJPhone - everywhere was a problem with iLBC. Where could be the problem? How in FreeSWITCH send to clientB iLBC payload type that incoming from clientA without enabled proxy_media? Thanks. -- Alex Gusak

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clustered freeswitch
by Vieri 24 Jun '08

24 Jun '08

Hi. I'm completely new to freeswitch and would like to migrate from asterisk. I have a couple of questions though: 1) I understand I won't be able to use my B410P Digium ISDN (mISDN) BRI card yet, right? 2) my TE120P Digium PRI card should work. 3) can freeswitch work in a master-master cluster? Example: FreeSWITCH-1 --- MySQL nbcluster tables --- FreeSWITCH-2 I suppose I can do this if I enable ODBC in FreeSWITCH. But will both FreeSWITCH boxes get "confused" if they use the "same" clustered database (active/active)? (by the way, does the built-in sqlite support clustered tables?) SIP softphones and ATA phones would register via DNS SRV (or custom method) to any one of the servers but only one (the idea is to load balance registrations). Say, extension 101 registers to FS-1 and 102 registers to FS-2. Could 101 contact 102 simply by looking it up the clustered database (eg. querying the table may yield something like sip:102@FS-2)? This would be similar to Asterisk 1.4+ Realtime SIP registrations. In Asterisk 1.2 I use IAX2 friend trunks between the 2 servers and DUNDi for the lookup. Would queues and ring groups also work in a "clustered" freeswitch system? In case of "queues", where is the caller position stored? Who's counting? FS-1 or FS-2? Could the counter be stored in the clustered SQL table so that both FreeSWITCH servers get it right in case calls come in on both servers? example: PSTN---PRI---FreeSWITCH-1 --- clustered SQL --- FreeSWITCH-2---PRI---PSTN Calls coming in from the PSTN reach both FS-1 and FS-2. I would like these calls to enter a "clustered" queue which bridges calls to operator extensions which can be registered on either server. The caller positions should be coherent across both servers. Has anyone done this already? Thanks for your feedback. Vieri

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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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