May 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ESL Command Not Found
by Antonio Teixeira 16 May '11

16 May '11

Hello I'm using ESL to Fork a call inside another , this is the commom example of User (A) pressing 1 and calling another party (B). I'm forking the call by opening an inbound socket and issuing the command api('originate',"{step=1,origination_caller_id_number=XXXX,fail_on_single_reject=true,ignore_early_media=true,originate_timeout=90,parked_call_uuid=%s}sofia/external/%s@2XXXXXXX&socket( 192.168.0.12:8040 sync full) All works great ( i get the call) and as you can see the call to B is redirected to the IVR Server The problem is if i do on the channel B ( By esl) uuid_getvar step XXXXXXXXXXXXXXXXXXXXXXXXXX I get Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: -ERR%20command%20not%20found Has anyone faced this ?? BTW using the python ESL: Best Regards Antonio Teixeira

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Re: [Freeswitch-users] setting up forward incoming calls to external gateway
by Kamen 15 May '11

15 May '11

Hi there! I still have that problem with pausing for 10 seconds when transferring an outside incoming call back to another external number. I would really really appreciate anyone with FS knowledge having a peek at it. I think I am giving enough information here. But if I am not, I am sure willing to provide more. However, my gutt feeling tells me the solution is easy and this is something silly I missed while configuring my FS. SO, here it goes, more info on the problem. I enabled debug info and here is what I see: 2011-05-15 18:14:12.187500 [DEBUG] switch_core_state_machine.c:157 sofia/external/9051212121(a)218.165.240.142 Standard EXECUTE EXECUTE sofia/external/9051212121(a)218.165.240.142 log(INFO dialing destination number) 2011-05-15 18:14:12.203125 [INFO] mod_dptools.c:1183 dialing destination number EXECUTE sofia/external/9051212121(a)218.165.240.142 bridge(sofia/gateway/mysipprovider/15191212121) 2011-05-15 18:14:12.203125 [NOTICE] switch_channel.c:812 New Channel sofia/external/15191212121 [55ef3edb-82b7-44dd-b961-3f6c33a5bd5e] So, new channel is created. And it looks like it is dialing. Then it goes 2011-05-15 18:14:12.203125 [DEBUG] mod_sofia.c:4286 (sofia/external/15191212121) State Change CS_NEW -> CS_INIT 2011-05-15 18:14:12.203125 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/15191212121 [BREAK] 2011-05-15 18:14:12.203125 [DEBUG] switch_core_state_machine.c:320 (sofia/external/15191212121) Running State Change CS_INIT 2011-05-15 18:14:12.203125 [DEBUG] switch_core_state_machine.c:356 (sofia/external/15191212121) State INIT 2011-05-15 18:14:12.203125 [DEBUG] mod_sofia.c:84 sofia/external/15191212121 SOFIA INIT An that was the last statement at the time of creating a channel. It pauses for 5 seconds and gives me: 2011-05-15 18:14:17.203125 [DEBUG] switch_nat.c:502 mapped public port 31306 protocol UDP to localport 31306 Well, ok, not sure why it needed 5 seconds for that. And then it pauses for another 5 seconds and eventually goes to the busy signal with the following debug log. 2011-05-15 18:14:22.203125 [DEBUG] switch_nat.c:502 mapped public port 31307 protocol UDP to localport 31307 2011-05-15 18:14:22.203125 [DEBUG] mod_sofia.c:124 (sofia/external/15191212121)State Change CS_INIT -> CS_ROUTING 2011-05-15 18:14:22.203125 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/15191212121 [BREAK] 2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:356 (sofia/external/15191212121) State INIT going to sleep 2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:320 (sofia/external/15191212121) Running State Change CS_ROUTING 2011-05-15 18:14:22.203125 [DEBUG] switch_channel.c:1668 (sofia/external/15191212121) Callstate Change DOWN -> RINGING hm... interesting. Why does it go to RINGING after 10 seconds pause? Thats the main question. 2011-05-15 18:14:22.203125 [DEBUG] sofia.c:4744 Channel sofia/external/15191212121 entering state [calling][0] 2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:359 (sofia/external/15191212121) State ROUTING 2011-05-15 18:14:22.203125 [DEBUG] mod_sofia.c:147 sofia/external/15191212121 SOFIA ROUTING 2011-05-15 18:14:22.203125 [DEBUG] switch_ivr_originate.c:66 (sofia/external/15191212121) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2011-05-15 18:14:22.203125 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/15191212121 [BREAK] 2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:359 (sofia/external/15191212121) State ROUTING going to sleep 2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:320 (sofia/external/15191212121) Running State Change CS_CONSUME_MEDIA 2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:378 (sofia/external/15191212121) State CONSUME_MEDIA 2011-05-15 18:14:22.203125 [DEBUG] switch_core_state_machine.c:378 (sofia/external/15191212121) State CONSUME_MEDIA going to sleep So, this is briefly the debug trace. I really hope anybody might tell me what is going on in here. Why sofia goes into that pause mode and tries to ring after 10 seconds, after actually it is disconnected. Most likely this has to do with my config. But I am being a total novice to FS have no idea what possibly can go wrong. Thanks in advance. Kamen -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/setting-up-forward-incoming-c… Sent from the freeswitch-users mailing list archive at Nabble.com.

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call waiting problem
by covici＠ccs.covici.com 15 May '11

15 May '11

Hi. I am using a Digium card for a phone extension. Now when someone calls that extension through my IVR they hear a nice ring after selecting the menu option. However, if I am on another call and they do this, they get silence. I do hear the call waiting beep, but why don't they hear the ringback or transfer_ringback? I tried it myself and indeed get silence. Thanks in advance for any suggestions. I do have a log if that would help. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Skypopen: start_skype_clients script permissions
by Chris Monkey 15 May '11

15 May '11

First, Giovanni, Thank You for the new skypopen installer, it really makes things easy... and stable. I am curious though, why must skype run as root? I'd prefer that it didn't, but I can't quite get it working when run as a normal user. I've given all rw permissions to /dev/dsp and the skype-clients-configuration-dir, which allows me to load mod_skypeopen and a skype call to ring in, but as soon as I answer, skypopen reports a hangup with normal clearing. Am I missing a permission somewhere else, or is there a fundamental reason why skype must be root for this to work? Thanks, Chris

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mod_freetdm not compiling on archlinux 2.6.32-lts kernel x86_64
by guru singh 14 May '11

14 May '11

Hi, Trying to install FS with mod_freetdm. Fails to compile. I have wanpipe installed Error : http://pastebin.freeswitch.org/16296 Can anybody with a working setup on this platform comment on this please? Thanks guru

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ESL: No events fired when there is error on submitted API command, like originate sofia to non-existent gateway
by Anton VG 14 May '11

14 May '11

I'm trying to catch an error, in case I would dial wrong (non existent) gateway I'm running ESL outbound listener, subscribing to all events, if I do proper bgapi 'originate' - there are normal events flow, and I can track what is happening. But if issue originate to a gateway, which is not configured - there is simply no any events fired. I have an error on FS console 2011-05-14 20:58:13.072927 [ERR] mod_sofia.c:4044 Invalid Gateway following by 2011-05-14 20:58:13.072927 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] But HOW to catch the given in ESL? Trying trick like checking for UUID exist or bgapi job ID - all unsuccessfull. sofia.c seems just does not have event code for that cases if (profile_name && !profile_found) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "No Such Profile '%s'\n", profile_name); >status = SWITCH_STATUS_FALSE; } Any clue?

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How to eliminate acoustic echo from server side(FreeSwitch)
by vivid 14 May '11

14 May '11

In the multi-party conference, because some termianl phone does not process echo, which causes the call have echo, so SERVER side needs to do echo processing. How to eliminate echo in this scenario? Do I need to add hareware device? If I want to implement it using software competely, what kind of open source echo library can be used? Does SPEEX work in Server Side? So many VIOP termianl use SPEEX, how to use it in the server side? Processing echo in server side should be different from client side, there are some complicated secnarios shuch as network delay and package lost, how to deal with these? Any advice will be appriciated.

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Domain tag in sofia profile
by Joseph L. Casale 14 May '11

14 May '11

I looked through the wiki but I couldn't find an explanation as to what you accomplish with the domain tags. When you alias a domain from the directory, what functionality does this provide? Thanks, jlc

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Different SAS with ZRTP
by Banhi Dutra 14 May '11

14 May '11

Hi all, I installed FS in test environment with ZRTP enabled. I have two softphone (I tested with Acrobits and Jitsi) registered and all works fine, except for SAS exchange, which are different. I'm using libzrtp 0.81.514 and FreeSWITCH Version 1.0.head (git-23d8658 2011-05-07 00-27-20 -0400). I read older post about a similar problem. Could be a version problem ? Any help will be appreciated Thank you Banhi

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problem with transferring calls when using ivr
by bogusmaster＠o2.pl 14 May '11

14 May '11

Hi all ! I'm a freeswitch newbie taking care of installation that I've recently taken over from a colleague who left our company. Currently I'm having a problem with transferring calls that went thru ivr menu. The call scenario looks like this: external number A calls our external number E. Then, in ivr menu it selects some option. After that A is connected with internal number I. When internal number I tries to transfer call from A to another internal number I2, call hangs up. If I turn the ivr off, as it is commented in the excerpt from /conf/dialplan/public/01_freephone_inbound.xml below, transferring calls works fine. <extension name="general_inbound"> <condition field="destination_number" expression="^(freephone_XXXX_in)$"> <action application="info"/> <action application="transfer" data="100 XML default"/>  </condition> </extension> Below is my ivr script: //==================================================================================== // Function :: Debug output //==================================================================================== function Debug(text) { var uuid_last5 = session.uuid.substr(-5) console_log("info", "["+uuid_last5+"] "+text+"\n"); } //==================================================================================== // Function :: Callback handlers for collectInput // Returning false from these handlers causes the digit collection // to "unblock". Depending on what behavior you want to achieve // you might want to unblock on any input or on a specific digit. //==================================================================================== function Get_DTMF_Stop_On_Any(session, type, data, arg) { Debug("DTMF: "+data.digit+""); arg.digits += data.digit; return false; } //===================================================================================================================== //===================================================================================================================== //===================================================================================================================== //===================================================================================================================== //==================================================================================== // Main Section //==================================================================================== var dtmf_digits = new Object(); session.answer(); Debug("Playing welcome message..."); session.streamFile("/opt/freeswitch/sounds/mlife/ivr/welcome.wav"); dtmf_digits.digits = ""; var repeat = 0; while (session.ready() && dtmf_digits.digits == "" && repeat < 3) { //------------------------------------------------------ // Flush digits, ask for menu selection //------------------------------------------------------ dtmf_digits.digits = ""; session.flushDigits(); Debug("Playing announcement..."); session.streamFile("/opt/freeswitch/sounds/mlife/ivr/ann.wav", Get_DTMF_Stop_On_Any, dtmf_digits); if(dtmf_digits.digits == "") { session.collectInput(Get_DTMF_Stop_On_Any, dtmf_digits, 3000); } Debug("Menu Choice Selected = [" + dtmf_digits.digits + "]"); if (dtmf_digits.digits == "") { repeat++; if (repeat > 2) { Debug("Max tries reached"); break; } } else { if (dtmf_digits.digits == "1") { session.execute("execute_extension", "100 XML default"); } else if (dtmf_digits.digits == "2") { session.execute("execute_extension", "200 XML default"); } else if (dtmf_digits.digits == "3") { session.execute("execute_extension", "401 XML default"); } else if (dtmf_digits.digits == "4") { //session.execute("transfer", "ivr-support"); Debug("Playing warning message about recording..."); session.streamFile("/opt/freeswitch/sounds/mlife/ivr/support.wav"); session.execute("execute_extension", "tech-support XML default"); } } } // User exceeded the ivr prompt limit, say bye-bye // session.streamFile("conference/mlife/8000/conf-goodbye.wav"); session.hangup(); Any hints would be greatly appreciated. Regards, Bart -- Using Opera's revolutionary email client: http://www.opera.com/mail/

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