June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

If you use mod_nibblebill, please upgrade to rev 13866
by Darren Schreiber 20 Jun '09

20 Jun '09

Hello folks, An important stability issue was identified in the FreeSWITCH core ODBC drivers when utilizing decimal / float columns in databases (at least MySQL anyway). This has an adverse effect on users of mod_nibblebill, as you are likely using floating point or decimal columns to track cash amounts. FreeSWITCH may periodically core dump when a floating point value is retrieved from the database due to this bug. Please upgrade to rev 13866 (or at least apply <http://fisheye.freeswitch.org/browse/FreeSWITCH/src/switch_odbc.c?r1=12152& r2=13866> the patch - it can be applied independently if you are on at least rev 12152). You do not need to update mod_nibblebill, as the bug is not actually in mod_nibblebill, it just happens to cause the right conditions to exist to exhibit this behavior. Background information is in FSCORE-384 <http://jira.freeswitch.org/browse/FSCORE-384> . Thanks to Tony for acting quickly on this patch. - Darren

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need to get some basics right
by xbipin 20 Jun '09

20 Jun '09

im just confused about certains tuff in freeswitch, firstly if we use TLS for SIP then FS can read such packets but if it were to act as gateway can it communicate in plain SIP, so its like phone connected to FS using TLS and then from FS to gateway in plain SIP? secondly, if phone uses SRTP, can FS take that media stream and convert to plain RTP so it can forward it to remote Gateway, i know it cant do for G723, G729 due to license issues but can it do for the other codecs? my main concern is can it actually act as a encryption and decryption server by remaining between the gateway and phone? -- View this message in context: http://www.nabble.com/need-to-get-some-basics-right-tp24122294p24122294.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Freeswitch as a B2B Application Server for IMS
by Cavalera Claudio Luigi 20 Jun '09

20 Jun '09

Hello freeswitchers, lately I'm trying to set up a testbed to ivestigate a potential use of freeswitch as a Back-to-Back application server in an IMS architecture. I've seen IMS specs are also linked here http://wiki.freeswitch.org/wiki/Documentation so I've thinked maybe there's a chance :-) I also have the inner feeling that fs could do an amazing job in IMS field as he does in NGN. For my testing I'm now using OpenIMSCore as control layer (where phones do register) and I'm trying to put fs on top of it as a b2b application server to provide services. I would like to share with you my experience to see if something could be done about this scenario (or if it's worth the trouble at least). Alice and Bob are two users registered to the IMS Core and they both have a profile for which originating and terminating INVITEs get triggered towards the fs application server. When Alice calls Bob the call setup would include three legs: 1) Alice -> PCSCF (orig) -> SCSCF (orig) -> FS (orig) 2) FS (orig) -> SCSCF (orig) -> ICSCF -> SCSCF (term) -> FS (term) 3) FS (term) -> SCSCF (term) -> PCSCF(term) -> Bob This partially works already with fs out of the box, but there are a still a few issues to be solved. When FS starts the brand new leg 2) as a B2B User Agent he should keep the Route: header in the SIP INVITE "almost the same" as the one he received in the leg 1) I see here two different issues: a) Getting the Route: header out of incoming invite in leg 1) b) Writing the proper Route: header and have FS behaving correctly at transport layer in the outgoing INVITE in leg 2) a) Now please correct me if I'm wrong: at the moment the <Route:> header is not a channel variable available in fs (e.g. I don't get it with the "info" app). It there were a way to get this header out of the incoming INVITEs, I could do the logic to parse it and forge a proper one in the outgoing INVITE. b) Concerning how to write the <Route> header, I'm already working with fs_path directive which also makes FS behaves correctly at network layer. Could someone please elaborate a little bit about the alternative to fs_path directive? I've seen there are already many in theory: - combining sip_h_Route= with http://wiki.freeswitch.org/wiki/Variable_sip_network_destination - use of http://wiki.freeswitch.org/wiki/Variable_sip_route_uri - use of fs_path= http://wiki.freeswitch.org/wiki/Sofia#Specifying_SIP_Proxy_With_fs_path I've simplified the scenario a little bit, there are other things that the B2B AS should do (e.g. removing Record-Route:) but FS do them already from what I've tested. If anyone in the community is interested I'm here to provide further information or share my experience to implement the best solution. Best regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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Mod_Fax / TxFax / Originate
by Tim B 20 Jun '09

20 Jun '09

Trying to do a local test for faxing. Keep getting an error. default dialplan: <extension name="test_rxfax_stream"> <condition field="destination_number" expression="^8000$"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="set" data="last_fax=${caller_id_number}-${strftime(%Y%m%d%H%M%S)}"/> <action application="rxfax" data="storage/fax/8000/inbox/${last_fax}.tif"/> <action application="hangup"/> </condition> </extension> //inbound from remote box works fine - connect asterisk box and fs box, then fax from asterisk to fs... OK - also fax from fs to asterisk.... OK // local fax on fs .... FAILS!! // my originate command: originate sofia/internal/8000(a)192.168.10.35 &txfax(storage/fax/test.tif) // error as follows: 2009-06-17 09:54:44.256749 [INFO] mod_dialplan_xml.c:252 Processing FreeSWITCH->8000 in context public 2009-06-17 09:54:44.256749 [ERR] sofia.c:4169 Cannot Blind Transfer 1 Legged calls 2009-06-17 09:54:44.256749 [NOTICE] sofia.c:3770 Hangup sofia/internal/8000(a)192.168.10.35 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] _________________________________________________________________ Lauren found her dream laptop. Find the PC that’s right for you. http://www.microsoft.com/windows/choosepc/?ocid=ftp_val_wl_290

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(Found Fix) Transmit fax locally for test
by Tim B 20 Jun '09

20 Jun '09

Ok so after many attempts of trial and error I narrowed it down to acls. So when trying to orginate a call to the local FS extension it was getting blocked. Adding the following allow with my freeswitch IP to the domains list allowed the originate to take place. acl.conf.xml: <list name="domains" default="deny"> <node type="allow" cidr="192.168.10.35/32"/> <node type="allow" domain="$${domain}"/> </list> So now this statement works for local fax testing: originate sofia/default/8000(a)192.168.10.35 &txfax(storage/fax/test.tif) Now my question is, is this the proper or best way to configure this? Tim > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 19 Jun 2009 10:00:35 -0500 > From: Michael Collins <msc(a)freeswitch.org> > Subject: [Freeswitch-users] Update - Transmit fax locally for test > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: > <87f2f3b90906190800u5d9436cbu2bd594bc8d09503(a)mail.gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Tim, > > Look at lines 47 and 48 of the pastebin. I think something goofy is > happening there. What is 8000(a)x.x.x.x in your system? Is that the receive > fax extension? > -MC > > ---------- Forwarded message ---------- > From: Tim B <timb0311(a)hotmail.com> > Date: Fri, Jun 19, 2009 at 7:39 AM > Subject: Re: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188 > To: freeswitch-users(a)lists.freeswitch.org > > > here is the log... > http://pastebin.freeswitch.org/9440 > > haha, yeah i see it now... duh. pulled an all nighter, too many things > going on. must have overlooked it. > > > > When I connect to pastebin.freeswitch.org I get a helpful notice saying > > the login and password is pastebin/freeswitch > > > > been trying to break myself into freeswitch on top of my original workload. > thanks for the help. > _________________________________________________________________ Bing™ brings you maps, menus, and reviews organized in one place. Try it now. http://www.bing.com/search?q=restaurants&form=MLOGEN&publ=WLHMTAG&crea=TEXT…

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Re: [Freeswitch-users] Update - Transmit fax locally for test
by Tim B 20 Jun '09

20 Jun '09

Yes that is the extension defined in the default dialplan. It is setup like explained under mod_fax... here is the actual definition: http://pastebin.freeswitch.org/9450 > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 19 Jun 2009 10:00:35 -0500 > From: Michael Collins <msc(a)freeswitch.org> > Subject: [Freeswitch-users] Update - Transmit fax locally for test > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: > <87f2f3b90906190800u5d9436cbu2bd594bc8d09503(a)mail.gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Tim, > > Look at lines 47 and 48 of the pastebin. I think something goofy is > happening there. What is 8000(a)x.x.x.x in your system? Is that the receive > fax extension? > -MC > > ---------- Forwarded message ---------- > From: Tim B <timb0311(a)hotmail.com> > Date: Fri, Jun 19, 2009 at 7:39 AM > Subject: Re: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188 > To: freeswitch-users(a)lists.freeswitch.org > > > here is the log... > http://pastebin.freeswitch.org/9440 > _________________________________________________________________ Hotmail® has ever-growing storage! Don’t worry about storage limits. http://windowslive.com/Tutorial/Hotmail/Storage?ocid=TXT_TAGLM_WL_HM_Tutori…

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high cpu utilization
by Nik Middleton 19 Jun '09

19 Jun '09

Hi Guys, This one has me a little baffled. If have a recent build (in the last week) of FS installed on two near identical HP servers. One happily runs 400 concurrent calls at around 50% CPU. The other can only run around 50 calls without the CPU going to 98%. Identical configs and lua script. Only diff is that the server having problems is running latest centos 64bit, where the other is 32bit. Any suggestions of where I might start looking? Regards,

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Diagnosing FreeSWITCH eating more and more memory
by Matthew Fong 19 Jun '09

19 Jun '09

With yesterday's trunk and also a release from 2 weeks ago, I noticed that my freeswitch process as it ran was eating up more and more memory. At the end of the day it was using 75% of the sever's memory (About 12 gigs). It starts out taking a small amount of memory, and then throughout the day it slowly takes more and more. Is this normal? I'm using several lua ivr scripts...and have about 600-900 channels. Whats the best way to go about tracking down the cause? Thanks. --matt

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CHANNEL_HANGUP event no longer having variable_billsec in header
by Matthew Fong 19 Jun '09

19 Jun '09

I upgraded to 13857 today, but noticed that the channel_hangup event no longer contain the variable_billsec header. Is this correct, or am I crazy? Thanks. --matt

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Controlling Conference Controls
by Bradley Brashier 19 Jun '09

19 Jun '09

How much power do I have with DTMF conference controls? The wiki doesn't have much information on this. For example, one of the things I'd like to do is take the currently existing "lock" and "unlock" actions and merge them into a "lock toggle" action. Preferably in XML configuration files. Is this even possible? If so, how would I get started? There are a variety of small things like this that I need to implement. Would I be better off switching to Lua?

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