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Call For Help: FreeSWITCH Cookbook
by Michael Collins 03 Jun '11

03 Jun '11

Hello FreeSWITCHers! We have a need for some assistance with a few of our recipes for the cookbook. If you are able to write technical documentation in English and have first-hand knowledge of these topics then you might be in a position to help: CDRs: Parsing XML CDRs Handling A and B leg CDRs Using a web server to handle XML CDRs Event Socket: Inbound ESL connections (basic how-to) Outbound ESL connections (basic how-to) Launch an outbound call with inbound event socket & ESL Handle inbound call with socket app & ESL Reacting to events (events for "my call" vs. system-wide events, etc.) Misc: Presence for BLF and SLA Presence for FIFO agent status If you think you've got the chops to write one or more recipes for the cookbook then email me off list and we'll discuss the specifics. I will get you all the information you need to get started. Thanks! -Michael

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Forcing SRTP/TLS connections
by Chris Cureau 02 Jun '11

02 Jun '11

Hi, everyone. I've begun experimenting with SRTP and TLS connections since the ZRTP download seems offline for the foreseeable future...sigh. I've got the SRTP enabled in my conf/vars.xml for both internal and external connections, but it appears that I can still connect without security from an unconfigured ATA. What I want to know is how to enforce a secure connection at registration time. Ideally, I'd like to do this both between two freeswitch servers and between a freeswitch server and an ATA. Thanks in advance!

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mod_msn
by Madovsky 02 Jun '11

02 Jun '11

Do you think it's technically feasible to make a module that connect to MSN network ?

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Re: [Freeswitch-users] Unable to get events from freeswitch using java ESL client
by Yungwei Chen 02 Jun '11

02 Jun '11

I missed the following: client.setEventSubscriptions("plain", "ALL"); From: freeswitch-users-bounces(a)lists.freeswitch.org [freeswitch-users-bounces(a)lists.freeswitch.org] Sent: Wednesday, June 01, 2011 6:28 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Unable to get events from freeswitch using java ESL client Hi, I am using org.freeswitch.esl.client-0.9.2 to ask freeswitch to make an outbound call in the following java class. I want to be able to get notified when a call is not answered for whatever reason. However, I don't get any events from freeswitch. What am I missing here? Thanks. public class esl implements IEslEventListener { /** * @param args */ public static void main(String[] args) { esl e = new esl(); Client client = new Client(); EslMessage response = null; try { client.connect("192.168.6.18", 8021, "ClueCon", 10); client.addEventListener(e); response = client.sendSyncApiCommand("originate", "{ignore_early_media=true}sofia/gateway/broadvoice/1112223333 9886"); CommandResponse cr = client.close(); System.out.println("Done."); } catch (InboundConnectionFailure ex) { ex.printStackTrace(); } } @Override public void backgroundJobResultReceived(EslEvent arg0) { System.out.println("backgroundJobResultReceived"); } @Override public void eventReceived(EslEvent arg0) { System.out.println(String.format("eventReceived, event name=%s", arg0.getEventName())); } }

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ClueCon 2011 News and Notes
by marketing＠cluecon.com 02 Jun '11

02 Jun '11

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this gona be fun?
by Seven Du 02 Jun '11

02 Jun '11

http://sites.google.com/site/webrtc/faq also, does that means iSAC available freely? -- Seven Du About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow

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FS vs provider's Sonus DTMF (double digit) problem
by Dmitry Saratsky 01 Jun '11

01 Jun '11

Hi All, I'm suffering from dtmf "double digit" problem for incoming calls that destined to IVR on my FS. I tryed to identify and workaround the provlem with no success... I have already tryed to change dtmf-mode, use sonus-fixup etc with no luck... I have: FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500) I would like to ask you some questions and I'm attaching some relevant output (see below) 1. According to output (see below), is it right to say, that my provider sends me DTMF events both in "DTMF INFO" and "RTP DTMF - rfc2833" way? 2. If so, may be this is the reason for "double digit" I suffer? 3. If so, can I filter incoming "DTMF INFO" events in FS? 4. If so, do FS have some workaround for this? Thanks, Dmitry. ------------------------------------------------- FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500) ------- <profile name="providers"> <gateways> <X-PRE-PROCESS cmd="include" data="providers/*.xml"/> </gateways> <aliases> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="shutdown-on-fail" value="true"/> <param name="auth-calls" value="false"/> <param name="sip-trace" value="no"/> <param name="disable-register" value="true"/> <param name="rfc2833-pt" value="101"/> <param name="dialplan" value="XML"/> <param name="context" value="providers"/> <param name="apply-inbound-acl" value="013SONUS-IPS"/> <param name="hold-music" value="$${hold_music}"/> <param name="stun-enabled" value="false"/> <param name="stun-auto-disable" value="true"/> <param name="manage-presence" value="false"/> <param name="nonce-ttl" value="60"/> <param name="dtmf-type" value="rfc2833"/> <param name="dtmf-duration" value="2000"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="50"/> <param name="rtp-hold-timeout-sec" value="300"/> <param name="sip-port" value="$${sip_port}"/> <param name="rtp-ip" value="$${rtp_ip}"/> <param name="sip-ip" value="$${ext_ip}"/> <param name="inbound-codec-prefs" value="PCMU,PCMA"/> <param name="outbound-codec-prefs" value="PCMU,PCMA"/> <param name="codec-prefs" value="PCMU,PCMA"/> <param name="inbound-late-negotiation" value="false"/> <param name="inbound-codec-negotiation" value="scrooge"/> <param name="disable-transcoding" value="false"/> </settings> </profile> ------- 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6236 INFO DTMF(0) 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6310 dispatched freeswitch event for INFO 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_play_say.c:1649 done playing file 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3 digits t/o 2000 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6207 Bad signal 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6310 dispatched freeswitch event for INFO 2011-06-01 10:21:30.966061 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 0:4360 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:390 digits '00' 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00] [/^(20[1-2])$/] [0] 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00] [/^[3-9*#]$/] [0] 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:574 IVR menu 'main_ivr_heb' caught invalid input '00' ------- 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6236 INFO DTMF(7) 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6310 dispatched freeswitch event for INFO 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_play_say.c:1649 done playing file 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3 digits t/o 2000 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6207 Bad signal 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6310 dispatched freeswitch event for INFO 2011-06-01 10:23:24.083260 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 7:2760 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:390 digits '77' 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77] [/^(20[1-2])$/] [0] 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77] [/^[3-9*#]$/] [0] 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:574 IVR menu 'main_ivr_heb' caught invalid input '77' 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16@8000hz 1 channels 20ms

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FS vs provider's Sonus DTMF (double digit) problem
by Dmitry Saratsky 01 Jun '11

01 Jun '11

Hi All, I'm suffering from dtmf "double digit" problem for incoming calls that destined to IVR on my FS. I tryed to identify and workaround the provlem with no success... I have already tryed to change dtmf-mode, use sonus-fixup etc with no luck... I have: FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500) I would like to ask you some questions and I'm attaching some relevant output (see below) 1. According to output (see below), is it right to say, that my provider sends me DTMF events both in "DTMF INFO" and "RTP DTMF - rfc2833" way? 2. If so, may be this is the reason for "double digit" I suffer? 3. If so, can I filter incoming "DTMF INFO" events in FS? 4. If so, do FS have some workaround for this? Any ideas? Thanks, Dmitry. ------------------------------------------------- FreeSWITCH Version 1.0.head (git-1086cba 2011-05-23 22-51-43 -0500) ------- <profile name="providers"> <gateways> <X-PRE-PROCESS cmd="include" data="providers/*.xml"/> </gateways> <aliases> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="shutdown-on-fail" value="true"/> <param name="auth-calls" value="false"/> <param name="sip-trace" value="no"/> <param name="disable-register" value="true"/> <param name="rfc2833-pt" value="101"/> <param name="dialplan" value="XML"/> <param name="context" value="providers"/> <param name="apply-inbound-acl" value="013SONUS-IPS"/> <param name="hold-music" value="$${hold_music}"/> <param name="stun-enabled" value="false"/> <param name="stun-auto-disable" value="true"/> <param name="manage-presence" value="false"/> <param name="nonce-ttl" value="60"/> <param name="dtmf-type" value="rfc2833"/> <param name="dtmf-duration" value="2000"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="50"/> <param name="rtp-hold-timeout-sec" value="300"/> <param name="sip-port" value="$${sip_port}"/> <param name="rtp-ip" value="$${rtp_ip}"/> <param name="sip-ip" value="$${ext_ip}"/> <param name="inbound-codec-prefs" value="PCMU,PCMA"/> <param name="outbound-codec-prefs" value="PCMU,PCMA"/> <param name="codec-prefs" value="PCMU,PCMA"/> <param name="inbound-late-negotiation" value="false"/> <param name="inbound-codec-negotiation" value="scrooge"/> <param name="disable-transcoding" value="false"/> </settings> </profile> ------- 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6236 INFO DTMF(0) 2011-06-01 10:21:30.585146 [DEBUG] sofia.c:6310 dispatched freeswitch event for INFO 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_play_say.c:1649 done playing file 2011-06-01 10:21:30.604196 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3 digits t/o 2000 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6207 Bad signal 2011-06-01 10:21:30.845091 [DEBUG] sofia.c:6310 dispatched freeswitch event for INFO 2011-06-01 10:21:30.966061 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 0:4360 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:390 digits '00' 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00] [/^(20[1-2])$/] [0] 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:484 action regex [00] [/^[3-9*#]$/] [0] 2011-06-01 10:21:32.976879 [DEBUG] switch_ivr_menu.c:574 IVR menu 'main_ivr_heb' caught invalid input '00' ------- 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6236 INFO DTMF(7) 2011-06-01 10:23:23.902331 [DEBUG] sofia.c:6310 dispatched freeswitch event for INFO 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_play_say.c:1649 done playing file 2011-06-01 10:23:23.922329 [DEBUG] switch_ivr_menu.c:343 waiting for 2/3 digits t/o 2000 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6207 Bad signal 2011-06-01 10:23:23.942304 [DEBUG] sofia.c:6310 dispatched freeswitch event for INFO 2011-06-01 10:23:24.083260 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 7:2760 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:390 digits '77' 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77] [/^(20[1-2])$/] [0] 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:484 action regex [77] [/^[3-9*#]$/] [0] 2011-06-01 10:23:26.084811 [DEBUG] switch_ivr_menu.c:574 IVR menu 'main_ivr_heb' caught invalid input '77'

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Unable to get events from freeswitch using java ESL client
by Yungwei Chen 01 Jun '11

01 Jun '11

Hi, I am using org.freeswitch.esl.client-0.9.2 to ask freeswitch to make an outbound call in the following java class. I want to be able to get notified when a call is not answered for whatever reason. However, I don't get any events from freeswitch. What am I missing here? Thanks. public class esl implements IEslEventListener { /** * @param args */ public static void main(String[] args) { esl e = new esl(); Client client = new Client(); EslMessage response = null; try { client.connect("192.168.6.18", 8021, "ClueCon", 10); client.addEventListener(e); response = client.sendSyncApiCommand("originate", "{ignore_early_media=true}sofia/gateway/broadvoice/1112223333 9886"); CommandResponse cr = client.close(); System.out.println("Done."); } catch (InboundConnectionFailure ex) { ex.printStackTrace(); } } @Override public void backgroundJobResultReceived(EslEvent arg0) { System.out.println("backgroundJobResultReceived"); } @Override public void eventReceived(EslEvent arg0) { System.out.println(String.format("eventReceived, event name=%s", arg0.getEventName())); } }

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PLAY_AND_GET_DIGITS dialplann application: set <lang> and <say_type> to SAY subcommand
by Kozak Vladimir 01 Jun '11

01 Jun '11

Hello all. In example for "play_and_get_digits" dialplan app I see next extension. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits <extension name="play_and_get_digits with say"> <condition field="destination_number" expression="^(6500)$"> <action application="play_and_get_digits" data="1 1 1 3000 # say:'press one for technicial support' silence_stream://250 res \d+" /> </condition> </extension>This extension say somethin and get user DTMF. I need say digits and numbers on various (EN and ES) languages. It's possible to specify these vars for "say" dialplan application http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say. And it works correctly. Could you tell me How can I set language and say_type for tts_engine in "play_and_get_digits" app case? I try to use next extension, but it don't work (FS take 'es NUMBER 1234567' as <text_to_say> and say all this text on English): <extension name="read_extension3"> <condition field="destination_number" expression="^7890$"> <action application="answer"/> <action application="set" data="tts_engine=flite"/> <action application="set" data="tts_voice=kal"/> <action application="play_and_get_digits" data="1 1 1 3000 # say:'es NUMBER 1234567' /usr/local/freeswitch/sounds/starpound/silence.wav read_result \d+" /> <action application="event" data="Event-App-Type=READ-EXECUTED,Read-Result=${read_result}"/> <action application="log" data="Read-Result=${read_result}"/> <action application="park"/> </condition> </extension> Also I try to set vars: <action application="set" data="tts_engine=cepstral"/> <action application="set" data="tts_voice=maria"/>  and get failed result: EXECUTE sofia/internal/1007@172.26.200.250:5060 play_and_get_digits(1 1 1 3000 # say:'1234567' /usr/local/freeswitch/sounds/starpound/silence.wav read_result \d+) 2011-06-05 22:58:33.661014 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]! 2011-06-05 22:58:33.661014 [ERR] switch_ivr_play_say.c:2361 Invalid TTS module!

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