September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Not able to make call using external profile
by pankaj anand 21 Sep '09

21 Sep '09

hi folks, I m not able to make SIP calls using external profile. i have added the following lines to the $installdir/conf/dialplan/public.xml <extension name="echo"> <condition field="destination_number" expression="^9996$"> <action application="answer"/> <action application="echo"/> </condition> </extension> <extension name="public_extensions"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="bridge" data="sofia/external/$1@$${domain}"/> </condition> </extension> I m able to connect using 1000 and 1001 from public Internet. I am able to make an echo call. *when i type :* $: sofia status profile external reg It shows the list of the connected clients and their information. but when I m trying to make a call from one user to other user, it generates the following error 2009-09-18 15:41:45.675054 [NOTICE] switch_channel.c:602 New Channel sofia/external/1001(a)192.168.1.50 [fcb6c23e-bdcd-41dd-b73e-df07b71252be] 2009-09-18 15:41:45.677063 [INFO] mod_dialplan_xml.c:315 Processing 1000->1000 in context public 2009-09-18 15:41:45.679071 [NOTICE] switch_channel.c:602 New Channel sofia/external/1000(a)192.168.1.50 [1a537865-be53-42ce-b8f5-cc183f4f1306] 2009-09-18 15:41:45.688161 [ERR] sofia_reg.c:1568 No Matching gateway found 2009-09-18 15:41:45.688161 [NOTICE] sofia_reg.c:1588 Hangup sofia/external/ 1000(a)192.168.1.50 [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] 2009-09-18 15:41:45.688161 [INFO] mod_dptools.c:2093 Originate Failed. Cause: MANDATORY_IE_MISSING 2009-09-18 15:41:45.689090 [NOTICE] mod_dptools.c:2125 Hangup sofia/external/1001(a)192.168.1.50 [CS_EXECUTE] [MANDATORY_IE_MISSING] 2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/external/1001(a)192.168.1.50) Ended 2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1001(a)192.168.1.50 [CS_DESTROY] 2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/external/1000(a)192.168.1.50) Ended 2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1000(a)192.168.1.50 [CS_DESTROY] with regards Pankaj anand

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Any FreeSWITCH training courses out there?
by Gavin Henry 21 Sep '09

21 Sep '09

Hi all, Is there anyone out there doing beginner courses or conversion courses from an Asterisk mindset? Cheers. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com

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Call Tracing
by Klaus Teller 21 Sep '09

21 Sep '09

Hi, Say i have an inbound VoIP/SIP call that hits my FS box. Is it possible to to extract information about the intermediate hops that the call or the signaling went through? If so, what information can i get? Thanks, Gregoire. -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser

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Affordable GSM gateway for one cellphone?
by Fred-145 20 Sep '09

20 Sep '09

Hello I'm selling a basic solution for SOHO customers (FS is installed on their work computer running Windows or Macs) to handle an analog phone line. When they're on the road, in addition or instead of getting a notification by e-mail when someone calls their office, some users might want to have the Freeswitch server actually ring their cellphone so they can take calls. Besides taking a subscription with a VoIP provider that the Freeswitch server will use to ring their cellphone, I'd like to know what my options are when it comes to setting up a GSM gateway on the customer's premises, in case they don't want to depend on the Internet. Are there Freeswitch-compatible, affordable solutions to handle a single GSM subscription? I guess all it takes is having them take a second subscription with their GSM provider and inserting the SIM chip inside the gateway to have Freeswitch ring their cellphone, but I've never used those things. Thank you. -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Bind extention to a different Dialplan and cdr php?
by SP 20 Sep '09

20 Sep '09

you can use different php includes On Sunday, September 20, 2009, João Mesquita <jmesquita(a)freeswitch.org> wrote: > Frank, > > That kind of logic needs to be performed at your application, if I am not wrong. I will do some testing here, but I think that mod_xml_cURL sends purpose and other information regarding the type of data it is requesting so you can decide on your application exactly what to do. > > Regards, > > jmesquita > > On Sun, Sep 20, 2009 at 1:10 PM, Frank @ Impact <frank(a)impactfax.com <javascript:_e({}, 'cvml', 'frank(a)impactfax.com');>> wrote: > > > > > > > > > > > > > > > > > > > > > Currently all incoming calls to my FS to > all extensions are sent off by curl to a particular php > script called dialplan.php. > > I would like to have certain extensions > that are called to have their xml dialplan built by a > curl to a different php script, say dialplan2.php. > > > > Is there a way to have certain extensions > get their dialplan by calling a different php script other than the default? > > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins > Sent: Tuesday, September 08, 2009 > 3:37 AM > To: > freeswitch-users(a)lists.freeswitch.org > Subject: Re: [Freeswitch-users] > Bind extention to a different Dialplan andcdr php? > > > > > > > > On Sun, Sep 6, 2009 at 8:43 AM, Frank @ Impact <frank(a)impactfax.com <javascript:_e({}, 'cvml', 'frank(a)impactfax.com');>> wrote: > > > > > > > Is there a way to bind a particular extension to a > different dialplan php and a different cdr php script than the default one? > > > > > > > > > > Could you re-phrase this question with a bit more > detail? Thanks. > -MC > > > > > > > > > > > > _______________________________________________ > 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 > > > -- Shannon

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Bind extention to a different Dialplan and cdr php?
by Frank ＠ Impact 20 Sep '09

20 Sep '09

Is there a way to bind a particular extension to a different dialplan php and a different cdr php script than the default one?

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Unknown call drops.. INFO DTMF(3)
by Mark Campbell-Smith 20 Sep '09

20 Sep '09

Hi! I have just experienced some call drops and each time the sequence is the same in the freeswitch.log file. Both parties are sure that they did not accidentally hit the 3 button to send the DTMF tone (and the same thing has happened four times already after ~5 minutes). 2009-09-13 19:28:23.835216 [DEBUG] sofia.c:4329 INFO DTMF(3) 2009-09-13 19:28:23.835216 [DEBUG] sofia.c:4450 dispatched freeswitch event for INFO 2009-09-13 19:28:23.859408 [DEBUG] switch_rtp.c:1624 Send start packet for [3] ts=2591120 dur=160/160/13120 seq=64923 2009-09-13 19:28:23.879439 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=320/320/13120 seq=64924 2009-09-13 19:28:23.899455 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=480/480/13120 seq=64925 : : 2009-09-13 19:28:25.439404 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=12800/12800/13120 seq=65002 2009-09-13 19:28:25.459312 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=12960/12960/13120 seq=65003 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65004 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65005 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65006 2009-09-13 19:28:33.879341 [NOTICE] sofia.c:322 Hangup sofia/external/<number> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-09-13 19:28:33.879341 [DEBUG] switch_channel.c:1683 Send signal sofia/external/<number> [KILL] 2009-09-13 19:28:33.879341 [DEBUG] switch_core_session.c:932 Send signal sofia/external/<number> [BREAK] 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:371 sofia/external/<number> ending bridge by request from write function 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:426 sofia/internal_nat/1000(a)192.168.1.120 receive message [UNBRIDGE] 2009-09-13 19:28:33.900940 [DEBUG] switch_core_session.c:630 Send signal sofia/internal_nat/1000(a)192.168.1.120 [BREAK] 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:452 BRIDGE THREAD DONE [sofia/internal_nat/1000(a)192.168.1.120] 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:454 Send signal sofia/external/<number> [BREAK] 2009-09-13 19:28:33.912049 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal_nat/1000(a)192.168.1.120 [CS_EXECUTE] [NORMAL_CLEARING] Anyone have any idea what this sequence means and why I am getting this? Is it my sip provider or something in FreeSwitch? What does the 'Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65006' mean? Notice that dur (duration?) is increasing a lot until the call drops. Thanks!

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Trouble Getting session:getVariable("state") in Lua
by Matthew Fong 19 Sep '09

19 Sep '09

I'm having trouble getting the channel variable state in my Lua ivr example. I have tried both session:getVariable("state") session:getVariable("Channel-State") session:getVariable("answer_state") session:getVariable("Answer-State") but lua reports nil for all attempts I did a uuid_dump and it appears normal....and both Channel-State and Answer-State Variables are present...does anyone know why my Lua IVR can not get these channel variables? Thanks --matt uuid_dump:Event-Name: CHANNEL_DATA Core-UUID: ed5556a8-060f-4ce4-85bb-0a70b08120cf FreeSWITCH-Hostname: matthew-laptop FreeSWITCH-IPv4: 192.168.2.2 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-09-19%2012%3A47%3A20 Event-Date-GMT: Sat,%2019%20Sep%202009%2012%3A47%3A20%20GMT Event-Date-Timestamp: 1253364440904749 Event-Calling-File: mod_commands.c Event-Calling-Function: uuid_dump_function Event-Calling-Line-Number: 3298 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1001 Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 Call-Direction: outbound Presence-Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Caller-ID-Name: FreeSWITCH Caller-Caller-ID-Number: 0000000000 Caller-Network-Addr: 192.168.2.4 Caller-Destination-Number: 1001 Caller-Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 Caller-Source: src/switch_ivr_originate.c Caller-Context: default Caller-Channel-Name: sofia/internal/1001 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1253364439936068 Caller-Channel-Created-Time: 1253364439936068 Caller-Channel-Answered-Time: 1253364440900612 Caller-Channel-Progress-Time: 1253364439976071 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false variable_channel_name: sofia/internal/1001 variable_sip_local_url: 1001%40192.168.2.2 variable_sip_destination_url: %22user%22%20%3Csip%3A1001%40192.168.2.4%3A5061%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1001%2540192.168.2.4%253A5061%3E variable_is_outbound: true variable_ignore_early_media: true variable_originate_early_media: false variable_sip_nat_detected: true variable_sofia_profile_name: internal variable_sip_call_id: 690ad846-1fbd-122d-1599-0010c6ceb785 variable_sip_reply_host: 192.168.2.4 variable_sip_reply_port: 5061 variable_sip_user_agent: Linksys/PAP2T-5.1.6(LS) variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%201231630%201231630%20IN%20IP4%20192.168.2.4%0D%0As%3D-%0D%0Ac%3DIN%20IP4%20192.168.2.4%0D%0At%3D0%200%0D%0Am%3Daudio%2016406%20RTP/AVP%200%20100%20101%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A100%20NSE/8000%0D%0Aa%3Dfmtp%3A100%20192-193%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dptime%3A20%0D%0A variable_remote_media_ip: 192.168.2.4 variable_remote_media_port: 16406 variable_read_codec: PCMU variable_read_rate: 8000 variable_write_codec: PCMU variable_write_rate: 8000 variable_local_media_ip: 192.168.2.2 variable_local_media_port: 20442 variable_endpoint_disposition: ANSWER variable_current_application_data: api_epik_pocket.lua variable_current_application: lua

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Callback in Javascript, session.destroy() does not free the channel!
by Alberto Escudero 19 Sep '09

19 Sep '09

We are trying to create a callback application in Javascript. We get the callerid from the unanswered call and after destroying the session, we initiate a callback to the user to conenct it to a local extension in the dialplan. Although we have tried to destroy the first session, or even invoke a second script using apiExecute("jsrun",dialer.js"), tried session.hangup() or exit()... the first session does not seem to close properly until the whole chain of scripts are completed. Here is a piece of code that shows the concept (yes!, the sleep function is far from ideal. CPU loves it! ) function sleep(milliseconds) { var start = new Date().getTime(); for (var i = 0; i < 1e7; i++) { if ((new Date().getTime() - start) > milliseconds){ break; } } } if (session.ready()) { //We catch the caller_id caller_id_num = session.caller_id_num; console_log("Now we got your Caller ID\n"); //How long we want to wait to trigger a call back session.execute("sleep",5000); console_log("We have waited a while... time to create the callback\n"); //apiExecute("jsrun", "callback.js"); } //Destroy the session... session.destroy(); session=undefined; sleep(10000); //Preparing callback session2 = new Session('{ignore_early_media=true}celliax/interface1/600464646'); session2.setAutoHangup(false); session2.answer(); exit(); ++ Wisdom thoughts? -- Stopping junk mailers is good for the environment

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Zaptel Analog FXO Outbound Dialing
by Michael S Collins 18 Sep '09

18 Sep '09

Gang, Does anyone have a working analog setup? I've reproduced symptoms on two different systems, one with the Rhino 8 port card and one with the Nxtvox TDM400 clone. Symptoms are identical when dialing out on the FXO port. The outbound dialing established a connection with the called party but audio does not flow to the endpoint until approx 30 seconds after the call is answered. Digilord is having the same issue. Anyway, before I open a bug report I wanted some feedback from those who are using FXO analog ports. Please let me know of your experiences. Thanks! -MC Sent from my iPhone

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