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Process_cdr question
by Vito Andolini 04 Oct '08

04 Oct '08

Let's say I am programatically initiating two calls and then bridging them together. If I have the dialplan as originate sofia/example/300(a)foo.com <blocked::mailto:sofia/example/300@foo.com> &bridge( <blocked::mailto:sofia/example/400@bar.com> sofia/example/400(a)bar.com) and have the process_cdr set to "true" which is the default. I'd like to know how both cdr would look like... Obviously the "b" leg will be logged starting right after "400" have answered the call... Which is fine.... Now would the "a" leg be logged starting right after its bridged to the "b" OR right after it picks up the call (even though "b" is being called or ringing at that moment) The answer is important, because I am using VOIP for both calls, and the voip starts charging me soon as "a" picks up the phone, so i am wondering if I am able to get the same record into my cdr to process it or do I have to run some sort of magic? Thanks, Vito A.

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Re: [Freeswitch-users] Load test - performance not even matching Asterisk
by Jon Bruel 04 Oct '08

04 Oct '08

Hi all An update on the performance measurements: The measurements I have referred to earlier all involved an Asterisk as the call generator. Somehow this setup leads to extensive rtp bandwidth usage. Each channel used around 500 kbps. If a phone is entered into the loop, this is reduced to the expected 64 kbps. I have not found any reason for this, but it certainly fouls up the test, and I have changed the test setup. Further, and since the earlier tests, the network has been updated to a Gbits network. I have now made two new test: 1) Using WinSIP from Touchstone as a call generator. 2) Using the Asterisk as one component, and setting up a chain of calls which goes forth and back from the Asterisk and the FS. All call are started from a real phone, and after 100 loops, where the calls are answered and sent on by the dial plan, the calls are terminated by an tone (<action application="gentones" data="%(500000,0,400)"/>) in the FS. The two test show similar top-figures at similar loads. The first test would be my preferable, but it is limited to 50 calls due to the trial licence limitations. Using an external non-FS and non-Asterisk device will eliminate some uncertainties, that's why it would be preferred. The other test has been done with 600, 400 and 200 channels (300, 200 and 100 calls), and the results of the top command are: cpu sy ni id wa hi si total * 600 10 30 0 33 0 2 25 100 FS600 22 33 0 30 0 0 15 100 0 * 400 7 18 0 67 0 1 7 100 FS400 14 17 0 62 0 0 7 100 0 * 200 3 10 0 84 1 0 2 100 FS200 7 8 0 82 1 0 2 100 The results do not show significant differences between the capacity behaviour of the Asterisk (*) and the FS. The also show an expected interrupt load (si) proportional to the square of the call load. Still the FS does not really outperform the Asterisk - which I find disappointing. Any comments are welcome.

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Call limit in freeswitch?
by shehzad p 04 Oct '08

04 Oct '08

Hi all, I am testing freeswitch with sipp calls, but freeswitch is accepting 70 to 80 calls only, no more calls are processed at all. Is there any configuration settings behind this, that limits calls? Is there any settings or parameters for call limit in freeswitch, so we can extend the call limits. Thanks msp -- View this message in context: http://www.nabble.com/Call-limit-in-freeswitch--tp19809553p19809553.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Routing problem
by henkoegema 04 Oct '08

04 Oct '08

I'm using http://www.voxbone.com for incoming virtual PSTN calls to voxbone(a)oegema.com oegema.com is my (dyndns)domain and voxbone is the extension in my domain (my FS server) Based on the callerid number to route incoming PSTN calls, I've been using following in Asterisk: [default] exten => voxbone,1,Goto(incoming,s,1) [incoming] exten => s,1,NoOP(Time=${STRFTIME(${EPOCH},,%H)}.${STRFTIME(${EPOCH},,%M)}) exten => s,n,GotoIf($[${CALLERID(num)}=32476378xxx]?default,2004,1) exten => s,n,GotoIf($[${CALLERID(num)}=32486632xxx]?default,2005,1) exten => s,n,GotoIf($[${CALLERID(num)}=0476378xxx]?default,2000,1) ..... ..... exten => s,n,Goto(default,1000,1) So far so good. ======================================================================================= Now I want to do the same in FS. If I don't have an extension "voxbone" and only use the following two extensios: <extension name="Henk GSM"> <condition field="caller_id_number" expression="32476378xxx"> <action application="bridge" data="sofia/internal/2004%$${domain}"/> </condition> </extension> <extension name="Tobie GSM"> <condition field="caller_id_number" expression="^32486632xxx$"> <action application="bridge" data="sofia/internal/2005%$${domain}"/> </condition> </extension> then calls from 32476378xxx and from 32486632xxx are bridged correctly. All other PSTN numbers are not routed anywhere. Which is correct.(because "voxbone" don't exist) However, when I add extension "voxbone" <extension name="voxbone"> <condition field="destination_number" expression="^voxbone$"> <action application="bridge" data="sofia/internal/2000%$${domain}"/> <action application="hangup"/> </condition> </extension> then ALL calls are routed to 2000.(voxbone(a)oegema.com exist) So the extensions <extension name="Henk GSM"> and <extension name="Tobie GSM"> are ignored. It is however not possible to put multiple condition fields for "caller_id_number" inside the "voxbone" extension <condition field="caller_id_number" expression="32476378xxx"> and <condition field="caller_id_number" expression="^32486632xxx$"> (like in Asterisk [incoming] exten =>s, ) See also discussion thread: http://www.nabble.com/GotoIf-td19793170.html So I can't route my calls based on caller-id anymore. So I'm still faced with the problem how to a route calls based on caller-id, and if the called-id doesn't match, the calls should be routed to extension "voxbone". My apologies if the story is too long, but I didn't find another wy to explain. Henk -- View this message in context: http://www.nabble.com/Routing-problem-tp19812334p19812334.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Newbie Questions
by Vito Andolini 04 Oct '08

04 Oct '08

Hi All, I am familiar with Asterisk and doing some testing for my next project. I have had some difficulties Asterisk, and now researching FreeSwitch hoping that it has some out-of-the box answers for my questions. Basically I want to implement a Click 2 Call service. Very simple, user types in his/her number on a website, that number is called, after it's answered, the company number is called and they are bridged together. 1) Is there a way to communicate with FreeSwitch programatically and issue commands such as initiate calls etc ? (ver much like manager API in Asterisk) 2) If you initiate a call from the software and then once its answered call a 2nd number. How do you bridge them? 3) After the 2 numbers talk and hang up. How does your cdr look like? Do you have 2 cdr's that correspond to both calls or just 1 after both numbers are bridged together? This is one of the problems I can't solve with Asterisk as it generates only 1 cdr after the 2 calss are bridged. The reason for this request is, in case of a Click2Call service, you are charged for both calls by your SIP provider therefore you need to be able to track both calls for invoices/payments etc. 4) Is there a way to programatically know if a call has been asnwered or not and act based upon that. I understand the cdr contains that information. But what I want is, if the call is not answered maybe I can play a prerecorded message or take them to the voicemail or whatnot. So I need a way to do a flow-control based on if the call has been asnwered or not in the dialplan. Does that exist? If so can you point me to some resources? Thanks, Vito A.

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I have a newbie question about this list
by Geoff Love 03 Oct '08

03 Oct '08

Is this list specific to technical information about freeswitch or are we also discussing business discussions as well. Similar to the * biz list. Just looking for feedback. Kind regards, Geoff Love -- Geoff Love Sales Engineer GLove(a)Teliax.com 303-629-8304

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All day meeting.
by Brian West 03 Oct '08

03 Oct '08

Don't forget today is the day you can call into sip:888@conference.freeswitch.org or 213-799-1400 and ask questions and interact with Mike, Anthony and Myself and other FreeSWITCHer's that might happen to be in the conference too. So if you have questions or want to report any issues please feel free to drop by every friday from 9am Central to around 6pm /b

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TLS freeswitch
by paulo leonardo 03 Oct '08

03 Oct '08

Hi, has anybody managed to setup TLS? When I change tls to TRUE in internal.xml, then freeswitch doens't listen on any ports (5060/5061/5080). I need a litle help. Version: freeswitch 1.0.1 (compiled) OS: ubuntu kernel: 2.6.24-16

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Re: [Freeswitch-users] Load test - performance not even matching Asterisk
by Jon Bruel 03 Oct '08

03 Oct '08

I have made some further studies of the performance, and after I have removed and old PC running 10 Mb/s on the Ethernet, the performance has been drastically improved. So I tentatively think that there has been a bottleneck somehow in the combination of Ethernet switch and slow Ethernet cards. I'll test further and report back. At this stage, Anthony, I need more testing before you look at my system. If any others have had similar experiences, please report back. /Jon

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Shortest path to recording MP3
by Christian Jensen 03 Oct '08

03 Oct '08

Hi, I have a DID and it is working with FS. What I would like is for some sort of recipe to: 1. Get the phone to pick up 2. Play a file 3. Record until a key is pressed 4. Playback the recorded file 5. Re-Record if they hit any key 6. If they hang up any time during step 3 or after, I want the file to be stored as an MP3 7. Once the file is converted, hit a URL indicating a new file is available for pulling down. I can figure most of this out probably, but am feeling a bit lazy - just bullet points on the most direct route would be good. This is a proof of concept project. Thanks! Christian

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