November 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Unified logging of incoming mobile calls - partial solution for android
by Avi Marcus 20 Nov '11

20 Nov '11

I know many people run/use a PBX of some sort, and I figured I'd share what I discovered: I have a mobile phone that gets a lot of calls directly, but I wanted to log those calls to a CRM. This is a partial solution.. Android Notifier ( http://code.google.com/p/android-notifier/) is an app that can be installed on the android and send a UDP message to a server (or also to a desktop client) when calls come in. The desktop client can launch a website to pull data for a screen pop of some sort.... It seemed to sometimes lose sync (when wifi went off and then came back..) but it's got the last 10 messages or so. I have it send to UDP, and this small node.js script saves the time-stamped messages to a file. (Note: timestamp is in milliseconds..) //http://code.google.com/p/android-notifier/wiki/NotificationProtocol > //DEVICE_ID/NOTIFICATION_ID/EVENT_TYPE/EVENT_CONTENTS > var fs = require('fs'); > var log = fs.createWriteStream("incoming_call_log.txt", {'flags': 'a'} ); > var dgram = require("dgram"); > var server = dgram.createSocket("udp4"); > server.on("message", function (msg, rinfo) { > // console.log("server got: " + msg + " from " + rinfo.address + ":" + > rinfo.port); > // console.log("Got: " + Date.now() + "/" + rinfo.address +"/" + > rinfo.port + "/" + msg); > log.write(Date.now() + "/" + rinfo.address +"/" + rinfo.port > + "/" + msg + "\n"); > }); > server.on("listening", function () { > var address = server.address(); > console.log("server listening " + address.address + ":" + > address.port); > }); > server.bind(10600, "YOUR IP"); And you can background and disown it, so it stays running.. after you launch node.js. Due to the abnormal method of submitting - not HTTP - I didn't know how to write anything in another language to do this. I suppose you could do it with python twisted. And for a screenpop.. set the desktop notifier to launch: /home/folder/web.sh "deviceId={deviceId}&id={id}&type={type}&data={data}" web.sh contains: > #!/bin/bash > echo "$*"; #>> /tmp/websh.log; #logging... > var=$* > /usr/bin/chromium-browser "http://website.com/script.php?$var" Anyway, this is just the basics. Feel free to share how you integrate it. I didn't see the specs on how to decrypt the signal algorithimically, but it's part of the desktop app. -Avi Marcus

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Re: [Freeswitch-users] Timing issues in AWS? (Stephen Dame)
by Lon Baker 20 Nov '11

20 Nov '11

It would be interesting to compare timing issues across major could services. Amazon, Rackspace, linode, etc. I've had poor experiences under ESX and avoid virtualization for anything handling media. -- Lon -- Lon

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Dialplan help
by Elliott Vogel 20 Nov '11

20 Nov '11

Hello, How can I bridge and transfer a call at the same time. Basically I would like to ring an external number and a local extension at the same time, also if the external number doesn't pick up after 30 seconds stop I need to stop calling the external number and continue ringing local extension. How can I set this up in the dial plan? Elliott

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Authorization User
by David Villasmil 20 Nov '11

20 Nov '11

Hello Guys, When I create a GW it MUST have a username and password in the sofia config, so i set a generic one like "123" Problem is, i need to send out in the FROM the callerid of the actual caller... I've tried setting (lua) on the outgoing call: session:execute("set","sip_from_user=" .. caller_id .. "") session:execute("set","sip_auth_username=" .. caller_id .. "") session:execute("set","sip_req_user=" .. caller_id .. "") But still the only thing that is correctly set is: Remote-Party-ID: "David" <sip:8889990@1.2.3.4:5060 >;party=calling;screen=yes;privacy=off [Expert Info (Note/Undecoded): Unrecognised SIP header (Remote-Party-ID)] [Message: Unrecognised SIP header (Remote-Party-ID)] [Severity level: Note] [Group: Undecoded] From: "David" <sip:123@1.2.3.4:5060;transport=udp>;tag=jtr1yQHFUUp3H SIP Display info: "David" SIP from address: sip:123@1.2.3.4:5060 SIP from address User Part: 123 SIP from address Host Part: 1.2.3.4 SIP from address Host Port: 5060 SIP tag: jtr1yQHFUUp3H I would need to set the FROM correctly... Any thoughts? Thanks David

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Kernel timer, how does that affect FS?
by Henrik Aagaard Sørensen 20 Nov '11

20 Nov '11

Hi all. I just saw a post (in this mailing list) recently regarding the kernel timer in Linux and how it affects sound quality in FS. Can anyone help me understand what the kernel timer does, how it affects FS and how to solve a possible problem on Linux?

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Cannot receive CUSTOM event in Python
by Aasish Pappu 20 Nov '11

20 Nov '11

Hi All I am writing a python app using it with the flex rtmp client. The javascript function testevent( ) does fire an event that shows up on fs_cli as mod_rtmp receives it. How do I catch this event from python? I am using a input_callback method to handle events. The testevent never gets there, however DTMF events fired using the flex keypad do reach. What should I do? I also tried adding this line to my dialplan <action application="event" data="Event-Name=CUSTOM,Event-Subclass=rtmp::clientcustom"/> I found out the Subclass of this event by calling DUMP_EVENT from the sendevent function inside the mod/endpoints/mod_rtmp/rtmp_sig.c Thanks -- Aasish Pappu

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User Directory
by Elliott Vogel 20 Nov '11

20 Nov '11

I have been trying to put a user outside of a group like how it's shown on http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide but when I put a user outside of a group the user cant authenticate. Can anyone see what I'm doing wrong or is there error in the documentation? <include> <domain name="lyonl.com"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <variables> <variable name="default_areacode" value="312"/> </variables> <user id="1901"> <params> <param name="password" value="123456"/> </params> <variables> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> <variable name="toll_allow" value="611,domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="outbound_caller_id_name" value="LyonL Inc" /> <variable name="outbound_caller_id_number" value="+13127567000" /> </variables> </user> <groups> <group name="default"> <users> <user id="1900"> <params> <param name="password" value="123456"/> </params> <variables> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> <variable name="toll_allow" value="611,domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="outbound_caller_id_name" value="LyonL Inc" /> <variable name="outbound_caller_id_number" value="+13127567000" /> </variables> </user> <user id="1909"> <params> <param name="password" value="12345"/> </params> <variables> <variable name="toll_allow" value="611,domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="outbound_caller_id_name" value="LyonL Inc" /> <variable name="outbound_caller_id_number" value="+13127567000" /> </variables> </user> </users> </group> </groups> </domain> </include>

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Loading an IVR menu on the fly via URL
by RaviRaj Mulasa 20 Nov '11

20 Nov '11

Hi FreeSWITCH Enthusiastics/Gurus Requirement 1. Customers can build a custom IVR using a UI. 2. Back end code persists the IVR details in a DB. 3. A URL will be exposed and will return the a XML file in the format as follows <menu name="CUSTOMER_ID_ivr" greet-long="phrase:demo_ivr_main_menu" greet-short="phrase:demo_ivr_main_menu_short" invalid-sound="/mnt/callControlStorage/en/invalidSelection.wav" exit-sound="/mnt/callControlStorage/en/tollFreeForwardingDotCom.wav" tollFreeForwardingDotCom timeout ="10000" inter-digit-timeout="2000" max-failures="3" digit-len="4"> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888(a)conference.freeswitch.org"/> <entry action="menu-exec-app" digits="2" param="transfer 9996 XML default"/>  <entry action="menu-exec-app" digits="3" param="transfer 9999 XML default"/>  <entry action="menu-sub" digits="4" param="3107767546_ivr_submenu"/>  <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/>  <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML default"/>  <entry action="menu-top" digits="9"/>  </menu> 4. Load and start executing the IVR 'CUSTOMER_ID_ivr' from events socket on the fly via HTTP URI. Gave it a shot with current setup ivr.conf.xml <configuration name="ivr.conf" description="IVR menus"> <menus> <X-PRE-PROCESS cmd="include" data="/mnt/<<Folder Name for the NETWORK DIRVE mapped>>/ivr_menus/*.xml"/> </menus> </configuration> The ivr.conf.xml is auto loaded at startup of FreeSWITCH. To load any new IVR menus , we need to call API command reloadxml which reloads the whole XML configs in autload_configs folder. The reloadxml might take long time based on the number of XML files present in the folder "/mnt/<<Folder Name for the NETWORK DIRVE mapped>>/ivr_menus" Good /Hope to have for the IVR menu 1. Load /Unload IVR menus on the fly using API command(s) freeswitch@localhost> loadivr << URL to fetch the XML describing the IVR menu(s)>> freeswitch@localhost> unloadivr <<IVR menu name>> 2. From the event socket SendMsg <uuid> call-command: execute execute-app-name: ivr execute-app-arg: <<MAIN_MENU to start execution>>SPACE <<URL to fetch the XML describing the IVR menu(s)>> Educated/Wild Guess We might be able to play IVR menu on the fly by combining the mod_xml_curl and dialplan tool - ivr. Please let us know how can we achieve the functionality , we are more interested in the second approach(Event Socket) under Good /Hope to have for the IVR menu. Thanks RaviRaj.

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Open Source MSC.
by Henrik Aagaard Sørensen 19 Nov '11

19 Nov '11

Hi everyone. I do know that this is the mailing list for FreeSwitch. But as I'm struggling finding anything (on Google etc.) about any open source MSC (HLR/AUC/SMSC/MMSC etc.) I'm trying here. I want to use FreeSwitch in a MVNO setup with a operator to make a proof-of-concept idea. But I need the MSC handling SIGTRAN and acts as HLR/AUC etc. I've looked at http://www.openimscore.org/ and http://www.mobicents.org, but I'm struggling figuring out there place in such setup. This is just for the proof-of-concept, so scale-ability, stability etc. is not as big a priority. Can anyone help me in the right direction? All help is appreciated!

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FS compile error
by fieldpeak 19 Nov '11

19 Nov '11

Dear fridends, i copy a FS source code (*FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400*) from a 32bit PC to a 64bit PC, on 32bit PC it compiles well, but on 64bit PC, there were below errors when compiling, could anyone help advise? Thanks. making all mod_hash /usr/bin/ld: skipping incompatible ./libesl.a when searching for -lesl /usr/bin/ld: cannot find -lesl collect2: ld returned 1 exit status make[5]: *** [fs_cli] Error 1 make[4]: *** [/usr/local/src/freeswitch/libs/esl/libesl.so] Error 2 make[3]: *** [mod_hash-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 [root@freeswitch freeswitch]# -- Regards, Charles

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