July 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

simple originate / bridge js
by paparoga＠mailinator.com 08 Jul '09

08 Jul '09

First of all I apologise for my long post. In the meantime I reworked the simple js as suggested (adding the domain or using transfer instead of bridge) but yet the script doesn't work. Let me add some other info. I just cloned the '5000' demo_ivr and reworked a little to reach my target. Next I added:  <extension name="ivr_demo_alarm"> <condition field="destination_number" expression="^118$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr_alarm"/> </condition> </extension> into the default.xml dialplan. Now all is ok if I connect Ekiga to FreeSwitch as user 1001 (for example) and than I dial '118'. The IVR works fine. Next from the console: originate sofia/my.freeswitch.address/1001 118 I get my Ekiga Phone ringing and connecter to the ext. 118 (the ivr) That is the log: ========================================================= originate sofia/10.0.0.33/1001 118 2009-07-08 21:56:57 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 [7e660b20-6bf9-11de-b2fa-f3963e050c84] 2009-07-08 21:56:57 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1001! 2009-07-08 21:57:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 21:57:00 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/1001 to XML[118@default] API CALL [originate(sofia/10.0.0.33/1001 118)] output: +OK 7e660b20-6bf9-11de-b2fa-f3963e050c84 freeswitch@Linux61> 2009-07-08 21:57:00 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSWITCH->118 in context default 2009-07-08 21:57:02 [WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample rate doesn't match 2009-07-08 21:57:15 [WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample rate doesn't match 2009-07-08 21:57:21 [NOTICE] sofia.c:3279 sofia_handle_sip_i_state() Hangup sofia/internal/1001 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 33 (sofia/internal/1001) Ended 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] ==================================================== Using the js instead it looks like if the bridge commands looks for a REGISTERED user at ext. 118, and so it fails. Using the transfer option, as suggested, the Ekiga user (1001) get to be connected to the ext 118, bur the connection drops immediately after. ==================================================== freeswitch@Linux61> jsrun alarm.js API CALL [jsrun(alarm.js)] output: OK freeswitch@Linux61> 2009-07-08 22:03:47 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 [7267f936-6bfa-11de-b2fa-f3963e050c84] 2009-07-08 22:03:47 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1001! 2009-07-08 22:03:51 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 22:03:51 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/1001 to XML[118@default] 2009-07-08 22:03:51 [NOTICE] mod_spidermonkey.c:2994 session_destroy() Hangup sofia/internal/1001 [CS_ROUTING] [NORMAL_CLEARING] 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 35 (sofia/internal/1001) Ended 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] ================================================== May be I'm on the wrong way, but this is my target: 1) From the external Perl freeswitch interface call a simple js 2) Make this script dial a sip/pstn phone number at the assistance location 3) Connect the just dialled assistance location to the IVR at ext 118 and let the support people hear some info about the raising fault. Thanks in advance for any suggestion. Roberto

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LUA Event generation
by Nick Lemberger 08 Jul '09

08 Jul '09

When I create an even with mod_lua, I need to add the even variable name as the 1st argument too all of the event statements - this is not how it's listed in the examples in the Wiki, is this new and should I update the WIKI or am I doing something wrong? The way I needed to write it to get it to work looks like this: ie: http://pastebin.freeswitch.org/9654 The examples in the wiki are just missing the first argument for addHeader(), addBody() and fire(). ie: http://wiki.freeswitch.org/wiki/Mod_lua#Events -Nick

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Can't understand documentation
by Lars Zeb 08 Jul '09

08 Jul '09

On http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session, at the bottom of "Activating via DTMF", it states: The other party doesn't hear the DTMFs but maybe its comfort noisy is disappearing for a very short time. When that sip client *starts* a call the above dialplan forbids activating recording. Can someone explain what these two sentences mean? Thanks, Lars

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simple originate / bridge js
by paparoga＠mailinator.com 08 Jul '09

08 Jul '09

Hi all, I'm attempting to setup a simple alarm handling machine. It should be triggered by an external event, dial a phone number (depending on the alarm type), and play a few wav files indicating the failure happened. Using Free, up to now, I've created a simple IVR and connected it to my EXT. 118. Connecting a softphone to the ext. 1001 and dialing the ext 118 the IVR is ok. Also using the following Free command from the console all is ok: originate sofia/zz.xxx.200.29/1001 118 My softphone at ext 1001 get ringed and then connected to the IVR at ext. 118. I cannot get the same from a simple js. I tried: ===================================== session = new Session("sofia/zz.xxx.200.29/1001"); //session = new Session(); //session.originate(session, "sofia/zz.yyy.200.29/118"); session.execute("bridge", "sofia/default/118"); ===================================== and almost all possibles variations, but I'get this result: 2009-07-08 16:23:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 16:23:00 [WARNING] mod_sofia.c:2495 sofia_outgoing_channel() Cannot locate registered user 118@default 2009-07-08 16:23:00 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() Close Channel N/A [CS_NEW] How can I tell to the script that the EXT. 118 is an IVR and not a registered USER? By the way, I attempted also the 'transfer' function, but I get the following: 2009-07-08 16:35:43 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1001! 2009-07-08 16:35:50 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 16:35:50 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/1001 to XML[sofia/default/118@default] 2009-07-08 16:35:50 [NOTICE] mod_spidermonkey.c:2994 session_destroy() Hangup sofia/internal/1001 [CS_ROUTING] [NORMAL_CLEARING] 2009-07-08 16:35:50 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 8 (sofia/internal/1001) Ended 2009-07-08 16:35:50 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] Any suggestion? Regards Kowalsky

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Sangoma 108 and libpri problems - only distortion sound
by Peter P GMX 08 Jul '09

08 Jul '09

Hello, I installed a Sangoma A108 with openzap and libpri. Signalling (E1) works sometimes (inbound and outbound calls are connected) but not always. Sound is just distortion but connection is stable. 2 questions: 1) What is the best way to go with Sangoma? OpenZAP with libpri or without libpri? (I remember there are some timer problems in openzap when not using libpri but this might have changed) However Sangoma recommends OpenZAP on their wiki. 2) What might cause the distortion? I crosschecked the config files and had a look at the interrupts (<2k/sec). Seems to be ok. ACPI and APIC is turned on. Freeswitch starts successfully with all the channels enabled. "oz dump" shows D-Channel up. "oz libpri debug 1 all" shows debugging messages with no special warnings. Best regards Peter Some confs for 1 Channel: Wanpipe1.conf [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 11 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE = NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF = YES TDMV_HW_FAX_DETECT = YES [w1g1] ACTIVE_CH = ALL TDMV_HWEC = YES MTU = 80 openzap.conf [span wanpipe PRI_1] name => OpenZAP number => 1 trunk_type => e1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 openzap.conf.xml <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="1"/> <param name="hold-music" value="$${moh_uri}"/>   </settings> <libpri_spans> <span id="1" name="PRI_1"> <param name="node" value="cpe"/> <param name="switch" value="euroisdn"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="q921loglevel" value="debug"/> <param name="q931loglevel" value="debug"/> <param name="dialect" value="q931"/> <param name="l1" value="alaw"/> </span> </libpri_spans> </configuration>

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ClueCon 2009 Updates - July 7, 2009
by Michael Collins 08 Jul '09

08 Jul '09

Hello folks! We have a few more updates. First of all, if you haven't already heard, we've extended the early bird sign up to go through July 21. That's only two weeks away, so if you haven't already registered then please call us at 877.742.CLUE and we'll get you set up. Secondly, there are some updates on the ClueCon blog: http://cluecon.com/blog/1 The breakfast and lunch menus have been posted. (Subject to change, of course. :) Also, we have a synopsis for Irv Shapiro's talk, entitled "Cloud Telephony" on the latest blog post. Can't wait to see you all in Chicago! -Michael Collins http://www.cluecon.com 877.742.CLUE

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CallID
by Maxim Tsvetov 08 Jul '09

08 Jul '09

Hi All Can somebody explain me algorithm of assigning CallID to new calls in Freeswitch. Are they unique? If not - how frequent they become duplicated. Regards, Maxim Tsvetov

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How to get the hook state?
by Eli Hayun 08 Jul '09

08 Jul '09

Hi I am a newbie in FreeSwitch and my question is: When I am calling to an extension, how should I know in advance what is the hook status. I tried to find out a variable that can get me this information but with no success. any help?

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Can't mute SIP channel with "receiveonly" in SDP
by Artem Shiyanov 08 Jul '09

08 Jul '09

Hy all! With Asterisk I can mute SIP channel using re-INVITE with "a=receiveonly" in media description. But this feature doesn't work with Freeswitch. For sure, there is old good method: transfer both legs to the conference room where one leg is able to listen/talk, the other one - only to listen, but this is unwanted workaround for me.. So I wonder: is there any other (preferably through the SIP) way to "mute" given SIP channel with Freeswitch? Thanks for all, Artem Shiyanov

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Record_session cutting wav files
by Mariusz Kołodziejczyk 08 Jul '09

08 Jul '09

Hello I saw a strange behavior when i'm using record_session for outbound call. Recorded file is 24:20 time length, but in logs should have about 25:18. Here ma log: start recording about 2009-07-07 10:56:15 2009-07-07 10:56:15.108447 [DEBUG] mod_dptools.c:748 sofia/internal/1062 SET [czas]=[2009-07-07-10-56-15] - variable $czas = "2009-07-07-10-56-15" i use it in filename below EXECUTE sofia/internal/1062 record_session(/records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav) ..... stop recording 2009-07-07 11:21:33.291177 [DEBUG] switch_ivr_async.c:444 Stop recording file /records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav file should have about: 25:18 time length when i listen a file it's really cut My piece of dialplan: <condition field="destination_number" expression="^(\d{4,10})$"> <action application="set" data="RECORD_ANSWER_REQ=false"/>  <action application="set" data="dir=${strftime(%Y-%m-%d)}"/>  <action application="system" data="mkdir -p /records/${dir}"/>  <action application="set" data="ringback=${pl-ring}"/>  <action application="set" data="czas=${strftime(%Y-%m-%d-%H-%M-%S)}"/>  <action application="record_session" data="/records/${dir}/${uuid}.${caller_id_number}.$1.${czas}.out.ISDN.wav"/>  <action application="bridge" data="{origination_caller_id_number=${cti_gateway_number},effective_caller_id_number=${cti_gateway_number}}openzap/1/A/$1"/> </condition> freeswitch version: FreeSWITCH Version 1.0.trunk (14013) linux: Linux pbx2 2.6.26-1-686 #1 SMP Fri Mar 13 18:08:45 UTC 2009 i686 GNU/Linux Thanks

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