February 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ivr from mysql
by michal kalinowski 02 Mar '10

02 Mar '10

Hi, I need build ivr script/aplication which will take dynamically configuration from mysql db (ivr menu, prompts, etc.). My first idea is generate xml from python script. But it's not working properly. Anybody has some idea or have this aplication already done ? BR, Michał

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Re: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA Support
by Anthony Minessale 02 Mar '10

02 Mar '10

You have to set the param to sylantro to get that mode. Or configure your phone to use broadsoft. On Feb 17, 2010 7:54 AM, "Yehavi Bourvine" <yehavi.bourvine(a)gmail.com> wrote: I am trying to migrate from 1.0.5pre10 to the latest (FreeSWITCH Version 1.0.5-20100216-0400 (16659M)), and shared apearance stopped working... We have Polycom phones and I use the same configuration for both versions. With the old version I see that after the phone registers with FreeSwitch, the server subscribes to the phone for the extension; with the latest version this does not happen. Furthermore, the table sip_shared_appearance_dialogs is empty. I don't find anything I can change on the phone config (the line is already set to shared). Here is the relevant config from the sip profile: <param name="manage-presence" value="passive"/> (set to TRUE only on one profile). <param name="manage-shared-appearance" value="true"/> <param name="dbname" value="share_presence"/> <param name="presence-hosts" value="pbx-gr-backup.ip-phones.huji.ac.il "/> Any idea where to look? Thanks, __Yehavi: _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Detecting Energy Levels on a Channel with mod_lua
by Matthew Fong 02 Mar '10

02 Mar '10

Is there anyway to detect energy levels on a channel that is being controlled by mod_lua? Please point me in the right direction if there is. Thanks. --matt

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smg_prid not bridging the call
by lakshmanan ganapathy 01 Mar '10

01 Mar '10

Dear all, I'm having a A102 Sangoma hardware. I configured it with freeswitch. wanrouter status, says both the port as connected. My smg_prid version is Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System restart============= Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack Daemon = Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version: 1.54 = Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15 2010 = Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release: wanpipe-3.5.8.6 = Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision: 15288 = Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: =========================================== My freeswitch version is 16729. I started freeswitch. oz list +OK span: 1 (smg_prid) type: Sangoma (boost) chan_count: 60 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none I originated a call as originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile. But when I issued the following command: originate openzap/smg_prid/a/9952248266 &bridge(openzap/smg_prid/a/8122133885) It rings my mobile (9952248266) first, but after that the following error was displayed 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION] The call got ended in my mobile. Freeswitch log and smg_pri.conf http://pastebin.freeswitch.org/12248 openzap.conf: [span wanpipe smg_prid] name => smg_prid trunk_type =>e1 b-channel => 1:1-15 b-channel => 1:17-31 trunk_type =>e1 b-channel => 2:1-15 b-channel => 2:17-31 openzap.conf.xml: <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <boost_spans> <span id="smg_prid"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </boost_spans> </configuration> Please guide me to setup this one!!.

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SQL ERR: library routine called out of sequence
by Yehavi Bourvine 28 Feb '10

28 Feb '10

Hello, After upgrading from 1.0.5pre10 to 1.0.5-20100216-0400 we started getting the above errors (I append bellow two samples). It seems Freeswitch fails to read a database using Sqlite. Anyone have seen this? Other details: Fedora 10, SQlite 3.5.9. We also do SQLite quesries during call setup via LUA from CoreDB. Is it an SQLite problem? Thanks! __Yehavi: The samples: 2010-02-16 16:22:13.762679 [ERR] switch_core_sqldb.c:404 SQL ERR [library routin e called out of sequence] delete from sip_dialogs where call_id=' 8656841832142-120172129116107(a)10.64.1.2' 2010-02-16 16:42:00.802442 [ERR] switch_core_sqldb.c:722 SQL ERR: [select call_i d,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_ho st,profile_name,hostname,network_ip,network_port,sip_username,sip_realm,mwi_user ,mwi_host from sip_registrations where profile_name='phones' and contact like '% 80635%'] library routine called out of sequence

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Phrases - Can't find macro
by Maciej Bylica 28 Feb '10

28 Feb '10

Hello, I am playing around with Phrases to use them with conference application. But i've encountered rudimentary problem of how to use newly added macro. What I already did is (according to wiki Speech Phrase Management) http://wiki.freeswitch.org/wiki/Speech_Phrase_Management - confirmed that there is "mod_say_en" loaded (<load module="mod_say_en"/>) - confirmed that there are proper .wav files - modified onf/lang/en/en.xml file: <section name="phrases" description="Speech Phrase Management"> <macros> <language name="en" sound_path="$${sounds_dir}/en/us/callie" tts_engine="cepstral" tts_voice="callie"> <macro name="confwelcome"> <input pattern="(.*)"> <match> <action function="play-file" data="vm-youhave.wav"/>  </match> </language> </macros> </section> - modified a part of dialplan: <extension name="conference_48112223344"> <condition field="destination_number" expression="^(\d{11})$"> <action application="phrase" data="confwelcome"/> </condition> </extension> I have following outcome: 2010-02-26 19:57:09.487245 [DEBUG] switch_core_session.c:638 Send signal sofia/internal/1000(a)217.153.192.36 [BREAK] EXECUTE sofia/internal/1000(a)217.153.192.36 phrase(confwelcome) 2010-02-26 19:57:09.487245 [DEBUG] mod_dptools.c:1850 Execute confwelcome() lang 2010-02-26 19:57:09.487245 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2010-02-26 19:57:09.496322 [ERR] switch_ivr_play_say.c:202 Can't find macro confwelcome. 2010-02-26 19:57:09.496322 [WARNING] switch_ivr_play_say.c:368 Macro [confwelcome] did not match any patterns Strange is that if i use a wiki example, it works. To be more precise: - conf/lang/en/en.xml file was overwritten with an example macros (directly from aforementioned wiki). - dialplan was modified: <extension name="conference_48112223344"> <condition field="destination_number" expression="^(\d{11})$"> <action application="phrase" data="msgcount"/> </condition> </extension> Am i doing something wrong? Thx, Maciej.

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end call detect on FXO port
by Brian May 27 Feb '10

27 Feb '10

Hello, I need to get Freeswitch to detect when the caller has hang up, so it will hang up the ilne also. Especially important for when the caller has left a message, although ideally it should work for all calls. With Asterisk this required the use of automatically detecting the busy signal, at the driver level. How can I do something similar with Freeswitch? Some web pages suggest I use this: <action application="tone_detect" data="busy 425 r 0 hangup 'normal_clearing' 3"/> Unfortunately this has the side affect that it answers the call, I don't want to change the answer behaviour, only the hang up behaviour. 2010-02-27 14:27:52.655081 [DEBUG] switch_core_session.c:1728 Application tone_detect Requires media! pre_answering channel OpenZAP/3:1/0397551926 I also see this message, but so far no solution: http://lists.freeswitch.org/pipermail/freeswitch-users/2008-July/004567.html How do I do this? Thanks -- Brian May <brian(a)microcomaustralia.com.au>

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ASR Apps
by Brian West 26 Feb '10

26 Feb '10

I'm looking for some enterprising community members to create some interesting voice apps using ASR. Please email me off list and we'll get you what you need to do this. Thanks, Brian

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Cannot make calls through PRI via wanpipe running as daemon
by Robert Hadley 26 Feb '10

26 Feb '10

When running Freeswitch as service called teoswitch as user teoswitch I cannot make calls through the Sangoma PRI or analog cards using wanpipe driver. I have added a file called 30-wanpipe.rules to /etc/udev/rules.d and rebooted the server. cat 30-wanpipe.rules # /etc/udev/rules.d/30-wanpipe.rules SUBSYSTEM=="wptdm", OWNER="teoswitch", GROUP="teoswitch", MODE="0660" SUBSYSTEM=="wanpipe", OWNER="teoswitch", GROUP="teoswitch", MODE="0660" Freeswitch log: Dialplan: sofia/internal/5410@192.168.72.45:5060 parsing [default->SangomaPRI] continue=false Dialplan: sofia/internal/5410@192.168.72.45:5060 Regex (PASS) [SangomaPRI] destination_number(93491045) =~ /^9(\d+)$/ break=on-false Dialplan: sofia/internal/5410@192.168.72.45:5060 Action set(effective_caller_id_number=425740${caller_id_number}) Dialplan: sofia/internal/5410@192.168.72.45:5060 Action bridge(openzap/smg_prid/a/3491045@g1) 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:122 (sofia/internal/5410@192.168.72.45:5060) State Change CS_ROUTING -> CS_EXECUTE 2010-02-25 16:51:11.328635 [DEBUG] switch_core_session.c:1019 Send signal sofia/internal/5410@192.168.72.45:5060 [BREAK] 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/5410@192.168.72.45:5060) State ROUTING going to sleep 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/5410@192.168.72.45:5060) Running State Change CS_EXECUTE 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/5410@192.168.72.45:5060) State EXECUTE 2010-02-25 16:51:11.328635 [DEBUG] mod_sofia.c:181 sofia/internal/5410@192.168.72.45:5060 SOFIA EXECUTE 2010-02-25 16:51:11.328635 [DEBUG] switch_core_state_machine.c:159 sofia/internal/5410@192.168.72.45:5060 Standard EXECUTE EXECUTE sofia/internal/5410@192.168.72.45:5060 set(effective_caller_id_number=4257405410) 2010-02-25 16:51:11.328635 [DEBUG] mod_dptools.c:811 sofia/internal/5410@192.168.72.45:5060 SET [effective_caller_id_number]=[4257405410] EXECUTE sofia/internal/5410@192.168.72.45:5060 bridge(openzap/smg_prid/a/3491045@g1) 2010-02-25 16:51:11.339637 [ERR] switch_core_session.c:357 Could not locate channel type openzap 2010-02-25 16:51:11.339637 [ERR] switch_ivr_originate.c:2411 Cannot create outgoing channel of type [openzap] cause: [CHAN_NOT_IMPLEMENTED] 2010-02-25 16:51:11.339637 [DEBUG] switch_ivr_originate.c:3209 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] 2010-02-25 16:51:11.339637 [INFO] mod_dptools.c:2346 Originate Failed. Cause: CHAN_NOT_IMPLEMENTED 2010-02-25 16:51:11.339637 [NOTICE] mod_dptools.c:2409 Hangup sofia/internal/5410@192.168.72.45:5060 [CS_EXECUTE] [CHAN_NOT_IMPLEMENTED] 2010-02-25 16:51:11.339637 [DEBUG] switch_channel.c:1976 Send signal sofia/internal/5410@192.168.72.45:5060 [KILL] 2010-02-25 16:51:11.339637 [DEBUG] switch_core_session.c:1019 Send signal sofia/internal/5410@192.168.72.45:5060 [BREAK] Thanks, Robert

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Conference - Originate Question
by Alfredo Quiroga-Villamil 26 Feb '10

26 Feb '10

All: I am currently using the following cmd to dynamically create a conference: originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf) I have noticed that when I send that cmd even if I specify: originate sofia/gateway/url/1XXXXXXXXXX &conference(myconf) public I am not hitting the dialplan. Is there a way to send the command and force it to hit the dialplan? Thanks in advance, Alfredo

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