February 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

RFC 4497 Originate Timeout / Progress Timeout .. No 100 Trying ... triggering 480 Response Code???
by Adam Long 12 Feb '09

12 Feb '09

Hi Guys, I've been experimenting with originate_timeout and progress_timeout as follows. However, shouldn't the timeout trigger a 408 Request Timeout instead of 480 Temporary Failure if no Provisional response received? Just curious, it seems to make sense to me.. but maybe SIP gods see differently. I have also tried using ${originate_disposition} after both bridge attempts to fetch the timeout disposition but instead this is set to NO_ANSWER (which would be correct for first attempt) As I understand it originate_disposition is reset for each bridge completed either successfully or unsuccessfully. Shouldn't the second attempt with no 100 Trying ever received trigger a NO_USER_REPONSE on timeout? According to RFC 4497 that would map to 408 Request Timeout For this test (please note progress_timeout set to low "2" value to test timeout) Node 10.200.1.11 is setup in such a way it responds with 100 Trying but never reaches 180 or 183 before 2 sec timer expires (as desired for this test) Node 10.200.1.12 (is disconnected and never even sends a provisional response, as desired) I have tried. <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="FSXML FastCgi Provided Dialplan"> <context name="public"> <extension name="FSXML.FCGI Generated Route"> <condition> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="call_timeout=120"/> <action application="set" data="progress_timeout=2"/> <action application="bridge" data="sofia/inside/+17001231234@10.200.1.11:5060"/> <action application="bridge" data="sofia/inside/+17001231234@10.200.1.12:5060"/> </condition> </extension> </context> </section> </document> As well as . <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="FSXML FastCgi Provided Dialplan"> <context name="public"> <extension name="FSXML.FCGI Generated Route"> <condition> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="{originate_timeout=120,progress_timeout=2}sofia/inside/+17001231234@10 .200.1.11:5060"/> <action application="bridge" data="{originate_timeout=120,progress_timeout=2}sofia/inside/+17001231234@10 .200.1.12:5060"/> </condition> </extension> </context> </section> </document> Any thoughts, am I completely nuts and missing something in the spec? Regards, -Adam

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Call accounting not working as expected
by Nik Middleton 12 Feb '09

12 Feb '09

I'm having an issue with call accounting If I initiate a call, and it is then transferred to an IVR menu. Person selects 1 to talk to someone. In js else if (data.digit == "5") { if (session.ready()) { var new_session = new Session(); new_session.originate(..... This Second call leg is not accounted for in either CSV or xml logs Am I doing something wrong? In the XML record is shows that I've diverted to the new number, but the time is all bundled with the initial call. This is exactly the same issue in Asterisk, which I was hoping to avoid. In Other words, why isn't a new call record created for the second leg? Regards,

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Change registration information for SIP-Registrar via XML/RPC?
by Rene Pankratz 12 Feb '09

12 Feb '09

Hello, we want to use mod_pa as a softphone, that registers to a SIPregistrar. But the username and password need to be changed over time without restarting freeswitch. Currently we are using XML/RPC to control the call functions. So it would be best (if possible) to use it also for changing registration information. Is there any way to do this? Thanks in advance René

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Switching from Asterisk to Freeswitch?
by Fred 12 Feb '09

12 Feb '09

Hello I successfully used Asterisk to build a voice server for our SOHO business. I did read the article comparing Asterisk to Freeswitch, but I have a couple of questions: 1. What are the decisive reasons that would justify taking a look at Freeswitch? What makes it a better option? 2. I'd like to build an affordable solution based on Asus' EeeBox and (because it's too small to add a PCI card) Sangoma's USB device to connect the host to POTS. Has someone successfully used Freeswitch to work on this hardware? www.asus.com/products.aspx?l1=24&l2=165 http://wiki.sangoma.com/sangoma-wanpipe-usbfxo Thank you for your feedback.

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Re: [Freeswitch-users] High CPU load after starting (Brian West)
by Public Dump 12 Feb '09

12 Feb '09

> OK does it work now? We have tested this on various windows installs > among the team here and not seeing this issue... it was a known issue > back in Nov. or Dec. but thats long been fixed. No, the problem is still there. I have tested it on a Core AMD 32bit AMD machine = everything is fine. On a 64bit 4 Core Intel Xeon machine = Problem is there.

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Socket-Event on "originate call"?
by Dennis 12 Feb '09

12 Feb '09

hi, i am using socket outbound with fs. if i do an originate over the console, for starting an outbound call without having an inbound call, and send the originate directly to the socket, the socket is first started, if the call is in answer or ringing state. before this, i will not receive any event, because the socket was not started. therefore i will not know, if the target is "busy" (hangup, hangup cause: user busy). it would be very helpful, if the socket would start immediately on an event like "channel originate". thanks for the help dennis

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stream a file multicast with mod_esf
by Sluschny, Thomas 12 Feb '09

12 Feb '09

Hi, i want to stream a file per IP multicast with mod_esf. I can stream IP multicast with: pa call stream XML and in XML dialplan: <extension name="stream"> <condition field="destination_number" expression="^stream$"> <action application="answer"/> <action application="esf_page_group" data="224.168.168.168 34567"/> <action application="esf_page_group" data=""/> </condition> </extension> and i can also play files with 'playback' app, BUT: how can put these 2 things together? May be its trivial, but i cant get it make working with 'originate' or 'uuid_broadcast' or 'bridge', 'transfer' and so on ... Thanks in advance, Thomas

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Re: [Freeswitch-users] High CPU load after starting
by Public Dump 12 Feb '09

12 Feb '09

http://files.freeswitch.org/freeswitch-snapshot.tar.gz Extracted into empty directory, SVN update, compile. > it was a completely clean build? > as in compleletely new and/or clean solution? > > Which tarball was it ? >

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gateways not hitting right context now?
by Brian West 12 Feb '09

12 Feb '09

If you have outbound gateways registering make sure you set the context and extension param on the gateway so it'll go to the right spot. Recent changes made it work much smoother. /b

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Skype as a path to wideband adoption
by Steve Underwood 12 Feb '09

12 Feb '09

Hi all, Consider the following: - FS has recently greatly enhanced its support for wide, wider and widest band telephony - That advantage is of no benefit when interworking with the PSTN - Skype has had wideband since day one, and just got super-ultra-duper-wideband. - FS is acquiring skype connectivity options (2, which hopefully will converge to one best in class option). One obvious conclusion is Skype offers the best possibility for having broad coverage with wideband voice. However, codec issues are certain to degrade quality. Lossy compressed codecs don't transcode well, and the codecs Skype uses are their own. FS is gaining a range of open and royalty free licence wideband codecs, but not the ones Skype uses. Skype choose widely supported narrowband codecs - G.711 and G.729 - but have not used a wideband codec with broad support. So.... the point of this note is "what can we do to optimise things?" Steve

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