September 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Nibblebill deduc balance anyway
by FERNANDO VILLARROEL 13 Sep '14

13 Sep '14

Dear All. When i do a call to a external number like my cellphone and If nibble_account have a balance equal to 0 the calls is tranfered to hangup context, it's works fine. But my mobile number ring and if I answer and then hangup nibblebill reduces the balance anyway I tried with ignore_early_media=true but not works <extension name="Todo_Chile"> <condition field="destination_number" expression="^([2-9]\d{8})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="lcr" data="$1 pg_prefix2"/> <action application="set" data="nibble_account=${accountcode}"/> <action application="set" data="nibble_rate=${lcr_rate_1}"/> <action application="set" data="nobal_amt=0"/> <action application="set" data="lowbal_amt=0"/> <action application="bridge" data=" {ignore_early_media=true,nibble_account=2,nibble_rate=35,nobal_amt=0,lowbal_amt=0}sofia/gateway/${lcr_carrier_1}/0$1"/> <extension name="hangup"> <condition field="destination_number" expression="^(hangup)$"> <action application="log" data="Limit de Credit Exceded !!!!!!!!"/>  <action application="hangup"/> </condition> </extension> I tried with lcr_auto_route but does not work too <action application="bridge" data="${lcr_auto_route}"/> As you can see : 2014-09-13 10:17:49.246372 [DEBUG] mod_nibblebill.c:563 Billing $2.193306 to 2 (Call: 381bd4f6-5ccf-4f67-a801-cfac03319639 / 0.000000 so far) 2014-09-13 10:17:49.246372 [DEBUG] mod_nibblebill.c:393 Doing update query [UPDATE auth_user SET saldo=saldo-2.193306 WHERE id='2'] my complete log http://pastebin.freeswitch.org/23282 What i doing wrong?

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high pdd
by Andre 12 Sep '14

12 Sep '14

What would cause very high pdd 12 seconds + after 3000 ports. Wireshark shows reinvites. I have tried many profiles and even many console apps both seem to have the same results after 2500 to 3000 ports (only 300 are connected) The server cpu and memory are ½ the server limits. Any ideas where to look would be great.

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 12 Sep '14

12 Sep '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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WebRTC and Third Party TURN Servers
by Lee Sylvester 12 Sep '14

12 Sep '14

Hi guys, So, the usage of WebRTC with FreeSWITCH looks pretty simple. What I’d like to know, though (and I can’t find it anywhere), is can I use a third party TURN server with FreeSWITCH? Thanks, Lee

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removing FS from SIP path after mod_fifo out
by George Niculae 12 Sep '14

12 Sep '14

Hi All, I am using mod_fifo for park/unpark like application using diaplan below, everything works great, I am able to park/unpark calls, transfer back to the extension that parked the call on timeout and remove FS from signaling path after transfer using uuid_simplify. What I wasn't able to accomplish is to remove FS from signaling path after unparking the call, tried both uuid_simplify and setting sip_auto_simplify variable without luck. Any idea if this is possible? Thanks, George <extension name="ParkOrbitFull"> <condition field="destination_number" expression="^5555$"> <action application="set" data="fifo_music=${hold_music"/> <action application="set" data="fifo_orbit_exten=transferBack${sip_referred_by_user}:30"/> <action application="set" data="fifo_orbit_dialplan=XML"/> <action application="set" data="fifo_caller_exit_key=5"/> <action application="set" data="fifo_caller_exit_to_orbit=true"/> <action application="fifo" data="QPark5555 in"/> <action application="hangup"/> </condition> </extension> <extension name="UnParkFull"> <condition field="destination_number" expression="^\*45555$"> <action application="fifo" data="QPark5555 out nowait"/> </condition> </extension> <extension name="transferBack"> <condition field="destination_number" expression="^transferBack(\d+)$"> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="{api_on_answer='uuid_simplify ${uuid}'}sofia/$${domain}/$1@$${domain}"/> </condition> </extension>

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Goip GSM Gateway works great with FreeSwitch!
by Dissident 12 Sep '14

12 Sep '14

http://freeswitch-users.2379917.n2.nabble.com/file/n6691087/single-channel-… Hi, I've configured this GSM gateway with Freeswitch and it works great. I'm posting this so that if somebody tries to configure this Gateway won't have to spend as much time as a I did :) This is the configuration: *Gateway* *freeswitch/conf/sip_profiles/external/goip.xml* *Tips to check that everything is OK - Remember to remove the PIN number from your SIM card. Double check your SIM card with a Phone. It should not ask you for a PIN. Disable it! Period :) - *sofia status profile internal* (the Goip extension user must appear, I chose ext. 1000 for it...) - * sofia status* (to check that the Goip gateway is properly configured. - On the Goip status page the Phone Status and your GSM Status must be LOGIN - On the Goip status page GSM operator GSM signal must show something. If it doesn't it's because it has not logged properly with your GSM Sim card. Double check it with a Phone. - If the Goip is logged properly with FS you can use your favorite SIP client to log directly into GOIP that is to say. if your GOIP IP is 192.168.2.2 you can log into 1000(a)192.168.2.2 *(again this is not your FS IP is the GOIP IP)* and make an outbound call to an ordinary cell phone. This is useful in case you have trouble with the outbound call. * Things to find out :?? - First I used X-Lite, Windows XP and miniSip server 20 clients to test this gateway. It's kind of strange that with miniSip every time I call to the extension I chose for Goip a tone appears an then I can dial a cell phone number. FS says that this extension is not available, it shows up with the sofia status profile internal though. *I've used with Linux Ubuntu: - FreeSWITCH Version 1.0.head (git-decfdbb 2011-08-11 14-15-26 -0500) - Jitsy Java Sip Client - 3 Android Phones with CSipSimple, Sipdroid and 3CXPhone Sip Client - 2 linksys spa941 - 1 Goip GSM GateWay http://goip.com.ua/download/goip-series-manual.pdf http://goip.com.ua/download/goip-series-manual.pdf http://www.alibaba.com/trade/search?SearchText=goip+gsm+gateway&Country=&In… http://www.alibaba.com/trade/search?SearchText=goip+gsm+gateway&Country=&In… I hope you find this info useful, if you have any comments or requests or corrections... please let me know... best regards. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Goip-GSM-Gateway-works-great-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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SkypOpen - Virtual Machine
by Alex Rixhardson 12 Sep '14

12 Sep '14

Hi Guys, I have installed Skypopen on a virtual machine, running Ubuntu 12.04 - 32bit. I can successfully make or receive Skype calls, but there is no sound at all. Are there any instructions how to set up audio for Skypopen that is running in a virtual machine? What I did so far was: + I checked that "lsmod |grep snd" returns nothing + I did run "cd /usr/src/freeswitch/src/mod/endpoints/mod_skypopen/oss" and "make clean; make; insmod ./skypopen.ko; mknod /dev/dsp c 14 3" before starting Skype and Freeswitch but that didn't help + I then loaded "modprobe snd-dummy" (I have snd-dummy installed), but that didn't help either I am using the latest Freeswitch (downloaded from git today) with skype-4.3.0.37 that is bundled together. Any help would be most appreciated. Regards, Alex

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group confirm not working after first bridge fails
by Grant Bagdasarian 12 Sep '14

12 Sep '14

Hello, I'm executing two bridge commands for failover, and I also have the group confirm enabled. It works perfectly fine for the first bridge. But when the first bridge does not answer, the 2nd bridge is executed. Once the 2nd bridge is answered (not confirmed), group confirm is executed, but when the destination presses the confirm key, nothing happens. Sometimes the DTMF digits don't even reach the FS server. Sometimes they do, but the application doesn't handle them. The call first hits the XML dialplan, then a js application is started. Has anyone experienced this before? First Bridge Command (From Console logs): bridge(<ignore_early_media=true,monitor_early_media_fail=user_busy:2:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true>[origination_caller_id_number=003100000000,origination_caller_id_name=003100000000,leg_timeout=15]sofia/outbound/0031612345678@192.168.1.1:5060<mailto:sofia/outbound/0031612345678@192.168.1.1:5060>) Failover Bridge Command (From Console logs): bridge([origination_caller_id_number=003100000000,origination_caller_id_name=003100000000,leg_timeout=15]sofia/outbound/031612345677@192.168.1.1:5060<mailto:sofia/outbound/031612345677@192.168.1.1:5060>) 2nd export of the group confirm variables (From Console logs): 2014-09-11 14:13:19.038451 [DEBUG] switch_channel.c:1180 sofia/inbound/003100000000(a)192.168.1.2 EXPORTING[export_vars] [group_confirm_cancel_timeout]=[1] to event 2014-09-11 14:13:19.038451 [DEBUG] switch_channel.c:1180 sofia/inbound/003100000000(a)192.168.1.2 EXPORTING[export_vars] [group_confirm_key]=[1] to event 2014-09-11 14:13:19.038451 [DEBUG] switch_channel.c:1180 sofia/inbound/003100000000(a)192.168.1.2 EXPORTING[export_vars] [group_confirm_file]=[/usr/src/freeswitch/sounds/accept_call.wav] to event Dialplan XML: <action application="set" data="continue_on_fail=true" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="bridge_early_media=true" /> JS Script: session.execute("export", "group_confirm_cancel_timeout=1"); session.execute("export", "group_confirm_key=1"); session.execute("export", "group_confirm_file=/usr/src/freeswitch/sounds/accept_call.wav"); session.execute("bridge", parallelBridgeCommand); if(failoverDestination != null) { session.execute("bridge", failoverDestinationCommand); }

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Skype is likely to stop supporting desktop API by Dec. 2013
by Henry Huang 12 Sep '14

12 Sep '14

Here is the link: http://www.engadget.com/2013/11/02/skype-killing-old-api/?ncid=rss_truncated I don't know if this mean that we'll no longer be able to interconnect with Skype anymore. And I just want to share the news with this community. Henry

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chat messages aren't being forwarded to the second user
by Snabel Kabiya 11 Sep '14

11 Sep '14

Hi, I'm using Linphone to test mod_sms on smart phones, the chat messages aren't being forwarded to the second user. I've two registered users 3333 and 4444. when i use action "send" the message is forwarded, but when i use my script with event fire it's not...what am i missing? *public.xml:* <?xml version="1.0" encoding="utf-8"?> <include> <context name="public"> <extension name="public"> <condition field="to" expression="^(.*)$"> <action application="lua" data="handle_chat.lua"/> <action application="reply" data="Hello, you said: ${_body}"/>  </condition> </extension> </context> </include> *handle_chat.lua* local body = message:getBody(); local from = message:getHeader("from"); local to = message:getHeader("to"); freeswitch.consoleLog("info", message:getHeader("from")); freeswitch.consoleLog("info", message:getHeader("to")); freeswitch.consoleLog("info", message:serialize()); freeswitch.consoleLog("info", message:getBody() .. "\n"); local event = freeswitch.Event("CUSTOM", "SMS::SEND_MESSAGE"); event:addHeader("proto", "sip"); event:addHeader("dest_proto", "sip"); event:addHeader("from", "sip:" .. from); event:addHeader("from_full", "sip:" .. from); event:addHeader("to", to); event:addHeader("type", "text/html"); event:addHeader("replying", "true"); event:addBody("Hello from Seven Du! Have funnnnnnn!"); freeswitch.consoleLog("info", event:serialize()); event:fire(); Thanks, Snabel

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