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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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Mod lcr and nibblebill problems
by Fernando Villarroel 11 Sep '14

11 Sep '14

Dear all. I am trying to use the both modules, but i have problems: Always the balance decreases and when i cancel the call the destination number ringing anyway. Also if the customer does not have balance, to try to make a call this goes smoothly in other words the calls always comes. The log http://pastebin.freeswitch.org/23267 My lcr.conf.xml <configuration name="lcr.conf" description="LCR Configuration"> <settings>  <param name="odbc-dsn" value="freeswitch:freeswitch:vmo123"/> </settings> <profiles> <profile name="default"> <param name="id" value="0"/> <param name="order_by" value="rate,quality,reliability"/> </profile> <profile name="qual_rel"> <param name="id" value="1"/> <param name="order_by" value="quality,reliability"/> </profile> <profile name="rel_qual"> <param name="id" value="2"/> <param name="order_by" value="reliability,quality"/> </profile> <profile name="pg_prefix2"> <param name="order_by" value="rate,quality,reliability"/> <param name="export_fields" value="nibble_account,nibble_rate,lowbal_amt,nobal_amt"/> <param name="custom_sql" value=" SELECT l.digits AS lcr_digits, c.id AS lcr_carrierid_field, c.carrier_name AS lcr_carrier_name, l.${lcr_rate_field} as lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix, a.id AS nibble_account, l.${lcr_rate_field} AS nibble_rate, a.warn_limit AS lowbal_amt, a.nobal_limit AS nobal_amt, a.id_tarifa AS tarifa FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id JOIN auth_user a ON a.id = ${accountcode} JOIN tarifas t ON t.id = a.id_tarifa WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND l.digits @> '${lcr_query_digits}' AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC, ${lcr_rate_field}, random(); "/> </profile> </profiles> </configuration> My Dialplan: <extension name="tests"> <condition field="destination_number" expression="^([23]\d{3})$"> <action application="lcr" data="$1 pg_prefix2"/> <action application="bridge" data="${lcr_auto_route}"/> </condition> </extension> What i doing wrong i am follow http://wiki.freeswitch.org/wiki/Mod_lcr#Integrating_with_mod_nibblebill I appreciate some help. regards Enviado desde mi iPhone

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How to know the event status when it is fired
by Steve Kieu 11 Sep '14

11 Sep '14

Hi, I am working on small chat application using lua script, basically grab all message details and make a custom event and fire. That works fine when the destination is online. But I want to know if the message is successfully sent in the script and if not then do something. How could I achieve that? Thanks in advance. -- Steve Kieu

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Git Update
by Andre 11 Sep '14

11 Sep '14

Hi, I've downloaded the Master a few weeks ago using git clone https://stash.freeswitch.org/scm/fs/freeswitch.git How do I get an update without having to download the full source again? Andre

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mod_lcr and nibblebill
by FERNANDO VILLARROEL 11 Sep '14

11 Sep '14

Dear all. I am trying to use the both modules, but i have problems: Always the balance decreases and when i cancel the call the destination number ringing anyway. Also if the customer does not have balance, to try to make a call this goes smoothly in other words the calls always comes. The log http://pastebin.freeswitch.org/23267 My lcr.conf.xml <configuration name="lcr.conf" description="LCR Configuration"> <settings>  <param name="odbc-dsn" value="freeswitch:freeswitch:vmo123"/> </settings> <profiles> <profile name="default"> <param name="id" value="0"/> <param name="order_by" value="rate,quality,reliability"/> </profile> <profile name="qual_rel"> <param name="id" value="1"/> <param name="order_by" value="quality,reliability"/> </profile> <profile name="rel_qual"> <param name="id" value="2"/> <param name="order_by" value="reliability,quality"/> </profile> <profile name="pg_prefix2"> <param name="order_by" value="rate,quality,reliability"/> <param name="export_fields" value="nibble_account,nibble_rate,lowbal_amt,nobal_amt"/> <param name="custom_sql" value=" SELECT l.digits AS lcr_digits, c.id AS lcr_carrierid_field, c.carrier_name AS lcr_carrier_name, l.${lcr_rate_field} as lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix, a.id AS nibble_account, l.${lcr_rate_field} AS nibble_rate, a.warn_limit AS lowbal_amt, a.nobal_limit AS nobal_amt, a.id_tarifa AS tarifa FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id JOIN auth_user a ON a.id = ${accountcode} JOIN tarifas t ON t.id = a.id_tarifa WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND l.digits @> '${lcr_query_digits}' AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC, ${lcr_rate_field}, random(); "/> </profile> </profiles> </configuration> My Dialplan: <extension name="tests"> <condition field="destination_number" expression="^([23]\d{3})$"> <action application="lcr" data="$1 pg_prefix2"/> <action application="bridge" data="${lcr_auto_route}"/> </condition> </extension> What i doing wrong i am follow http://wiki.freeswitch.org/wiki/Mod_lcr#Integrating_with_mod_nibblebill I appreciate some help. regards

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