- FreeSWITCH-users - FreeSWITCH Mailing Lists

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

2 1

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

2 6

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?

1 0

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

2 1

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>

2 1

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.

2 1

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!

8 11

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

1 0

rtp-timer-name issue and question
by Matt Stockton 19 Nov '11

19 Nov '11

Hi all, I am battling the following issue: My applications are primarily lua scripts that play files, accept DTMF , and make db calls and url callbacks to post information. At one point, I was having issues with an ever increasing delay for DTMF recognition as the application progressed (e.g. I would use playAndGetDigits, and the DTMF wouldn't be 'recognized' for a delayed amount of time. This time would increase and it appeared to be dependent on the number of db calls and url callbacks I made in the app -- as I made more, the delay would get worse). I asked around, and ended up changing this setting in my external sip profile: rtp-timer-name = none (it was soft before). This fixed my issue. Now, I am battling a separate issue, and I seem to have isolated it's cause to setting rtp-timer-name to none instead of soft. The problem now is that when I want to play an mp3 file, and I use streamFile with shout or playback with http_get, the beginning of the file is not played at all (playing starts probably about 200-400ms into the file. This is very repeatable for me - it's not happening to every streamed file, but it seems to be very deterministic on while line of lua script it's happening (the streamFile is happening in a loop, and it seems to always be happening to the first streamed file). If I change rtp-timer-name to soft, everything is fine, but then I'll have the first problem I described above. So, to fix one problem I need rtp-timer-name = none, for the other problem, I need rtp-timer-name = soft. I must be doing something wrong here, any suggestions? Thanks, Matt

1 2

Custom Call reporting
by Frank Busalacchi 19 Nov '11

19 Nov '11

Hello, and thanks for a great piece of software! I currently have FreeSwitch recording all inbound calls. I'm envisioning writing an intranet webpage that would allow managers to be able to generate a list of hyperlinks to call recordings based on ad hoc criteria such as an extension that was on the call path , CLID of the inbound call, date etc. I'm figuring the best way to do this is to name the inbound call recording by its UUID, and by parsing the logs and extracting the UUID of all calls that match the criteria. The system is currently configured to log through CDR-CSV with default installation settings. When I peruse the logs, I see only the beginning of the phone call (Initial inbound, or initial outbound)....What I don't see is the entire "call path". How do I get the log to show every transfer of the call etc? My thought is that if I could do that, then I could basically grab the UUID of the call, grab all log entries that have the UUID in it, and see every extension that was involved with that UUID. Am I on the right path? I'm tinkering with a production system, so I don't want to mess with settings and interrupt the business during business hours. My guess is that I need to set the CDR-CSV log setting to log both AB instead of the default to get the info I am looking for to show up in the log? Thanks, -Frank

2 1