September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Call Files for a dialer engine
by Alberto Escudero 23 Sep '09

23 Sep '09

I am exploring the possibility of building a Dialer that emulates the logic of Call Files in asterisk. A CallerID catcher is creating CUSTOM events that I can store in a database. I can trigger callbacks using ESL but I wonder what is the best way/nicer/geekier to do something like outgoing calls in * /aep -- Stopping junk mailers is good for the environment

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update on Windows installer
by Carlos Talbot 23 Sep '09

23 Sep '09

For those that are not aware, I've made some changes to the Windows installer over the past month. You can find a summary on this link: http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Precompiled… The latest change is the bundling of 32 and 64 bits builds in one file. The installer automatically selects the proper version based on the running OS. I encourage those who are using Windows to try out this release. (http://files.freeswitch.org/windows_installer/freeswitch.exe) I'm in the process of performance testing the 64 bit build on an HP DL380 w/ 16GB ram (8-way 3GHZ). As with the Linux build it's proven to be more scalabe than the 32 bit build on 64 bit hardware. I will follow up on wiki with my results. I'm trying to mimic this configuration: http://wiki.voiceworks.pl/display/~pawel/Maxed+out+CPS+test regards, Carlos note: it might take a while before you see this recent upload as the files server is synced via a content delivery system. When done you should see a file in that directory with today's date: LATEST_SVN_14953

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Automagic Phone Provisioning
by digilord 23 Sep '09

23 Sep '09

Hello all, I know this is done and I think I figured out how to do it but I don't want to reinvent the wheel so here goes. I am looking for a program that will sit on the PBX. This program will intercept DHCP reply packets destined for phones, inject "option 66" into the packet and release it back onto the network. Some of you might be wondering why I want a program like this. Simple. Lazy clients. They don't want to mess with their network infrastructure to assist us with automated deployment of SIP devices. They also don't want 50-100 devices connecting to an off site server downloading 20-40MB of firmware on a reboot. The PBX is not hard coded with an IP address. It's DHCP. They were willing to allow the PBX on the network and assign it a static DHCP address. Is what I am looking for not possible? Does someone have a sensible solution that doesn't involve dropping the client (yes someone suggested that)? Thanks in advance for any help you can give. DigiLord

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Mod_perl $session in not hangup
by lakshmanan 23 Sep '09

23 Sep '09

Hi all, I've the following mod_perl program to execute when I call to an extension (say 777). I use twinkle as a soft phone, to make calls. #!/usr/bin/perl use strict; use freeswitch; our $session; $session->answer(); if($session->ready()) { my $uuid=$session->getVariable("uuid"); freeswitch::consoleLog("INFO","UUID is $uuid\n"); freeswitch::consoleLog("INFO","Session is answered\n"); $session->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav"); my $dtmf = $session->getDigits(4,"", 5000); freeswitch::consoleLog("INFO","I received $dtmf\n"); $session->hangup("NORMAL_CLEARING"); sleep(5); # Some other statements. } return 1; Everything is fine. After executing $session->hangup, I got NORMAL_CLEARING in my freeswitch console. But in my soft phone, still the channel is active for 5 seconds. The call got ended only after the 5 seconds sleep. But if I create my own session like my $session=new freeswitch::Session("user/1000"); and I say $session->hangup(), it got terminated. I wanted to know why there is such difference?? or am I wrong?? Please clarify me. -- View this message in context: http://www.nabble.com/Mod_perl-%24session-in-not-hangup-tp25530646p25530646… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Using Sofia-SIP with ICE NAT Traversal Mechanism
by afshin afzali 23 Sep '09

23 Sep '09

Hi, I know that FreeSWITCH uses libdingaling to talk to Jingle call parties. Also I know that Jingle Protocol uses ICE protocol to traverse NAT devices. So : does libdingaling use an open library such as libnice for ICE? Is it possible to use the ICE implementation in Sofia-SIP endpoint? If not, how could I integrate an open ICE library in Sofia-SIP? Regards, -- afshin

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Hangup: Always the same Q.850 cause code
by Peter P GMX 23 Sep '09

23 Sep '09

Hello , I try to hangup aa call with a certain cause code. If the user dials a number which we currently do not serve we send <action application="set" data="sip_ignore_remote_cause=true"/> <action application="hangup" data="NO_ANSWER"/> which gives a "SIP/2.0 480 Temporarily Unavailable." Message , which is fine. For the target number being busy or having another state, we use this. <anti-action application="set" data="sip_ignore_remote_cause=true"/> <anti-action application="hangup" data="${hangup_cause}"/> which gives a "SIP/2.0 486 Busy Here."" Message , which is fine in case of Busy. However in both cases the SIP mssage has the following cause code: Reason: Q.850;cause=16;text="NORMAL_CLEARING". which can lead to problems when forwarding to a PSTN Gateway. How can we achieve, that the cause code is in sync with the deiivered message? We are on Trunk 14741M. Best regards Peter

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CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
by Michael Collins 23 Sep '09

23 Sep '09

Hello FreeSWITCHers! We are looking for people who are in a position to help out with various subprojects that will help FreeSWITCH to keep growing. We need people to help out in these basic areas: Bug marshals (people who watch JIRA and test bug reports, patches, etc.) Documentation maintainers (people who update the wiki when new stuff comes out, also those familiar with mediawiki administration) Documentation authors (people who write new docs, how-to's, tutorials, examples, etc.) Package maintainers (people who manage Debian debs, RPMs, etc.) Additionally, we are always looking for more folks to assist with answering questions on IRC and the mailing list. It is definitely nice to have people who've gone through the pains of switching to FreeSWITCH (or learning it from scratch) who can assist the steady stream of new users. If you want to help and aren't sure where to go from here then please at least do the following: #1 - Join #freeswitch on irc.freenode.net and hang out as much as possible #2 - Check the recent changes link on wiki.freeswitch.org each day #3 - Join the Friday public conference call and listen in These three things, in addition to the mailing list, will keep you well in tune with the FreeSWITCH community and what's happening. Next, make a note of the parts of FS that you use frequently, know a lot about, or are particularly passionate about. Those are the items we'd love to have you help us with. For example: if you use mod_xml_curl frequently and have been through the set up process then you're a prime candidate to help answer questions, refine the mod_xml_curl wiki documentation, write up a tutorial, contribute a working example of a web server & database schema, etc. If you are good with a scripting language then we could definitely use help with rounding out the docs for your favorite language. We could also use code samples, so ask for a contrib folder if you have things you would like to share. Or how about this: you read something on the wiki, it doesn't quite work when you try, so you tinker until you figure it out. Now you're in a position to update the wiki for everyone else's benefit, too. As you can see, you don't have to be a FreeSWITCH expert before you can help the project. What we really need are people who care about the project and want to see it flourish. If you are such a person then please contact me off list. Tell me what you're good at or where you would like to help. Many thanks for all of your support! -Michael

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Vestec Speech Recognition Integration ?
by Gerry Hull 23 Sep '09

23 Sep '09

Has anyone integrated Vestec Speech Recognition with FreeSwitch? It's $99/port... http://www.vestec.ca/ They have a C/C++ api, looks pretty simple. Alas, no MRCP until 2010.

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mod_nibblebill
by João Mesquita 23 Sep '09

23 Sep '09

Guys, I have been testing mod_nibblebill lately and there are 2 params that I could not make work.  <param name="percall_max_amt" value="1"/> <param name="percall_action" value="hangup"/> Looking at code, I could not find a single line that would actually test those. Is this confirmed to be implemented? If not, this should be removed from the configs so it won't get ppl lured. Regards, jmesquita

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Recording inbound call including DTMF - possible ?
by Morten Henckel 23 Sep '09

23 Sep '09

Hi I need to measure DTM digits duration and interdigit delay for various phones in a two stage dialing scenario. I.e Phone dials DID and after answer then the second number My set-up is: Phone->PSTN network->DID(inband DTMF) ->FS I ha ve FS to answer the call and record the call - all this is fine. However when i analyse the rdecording the Digits are being cut off down to 10 msec "bursts" - I trust its FS that cust the DTMF in order to avoid further propogation inband to second leg of the call. Is theer a way to avoid this ? I.e record the inbound call without DTMF processing ? Thx Morten

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