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How to bridge 2 sessions with Javascript?
by Nicolas Brenner 28 Mar '08

28 Mar '08

Hello everybody again, First of all I want to say thanks to the people on this list and on IRC, I'm really surprised (in a very good way) of the help I've received. Now to my problem: I'm trying to bridge two SIP calls together with a JS script, and to achieve it, I'm did the following: - created a dialplan entry for extension 500 which calls js script - created a js script with the following code: // Create new_session session1 = new Session(); session1.originate(session, "{ignore_early_media=true}sofia/gateway/sip.sipdiscount.com/005624949458"); session1.waitForAnswer(10000); new_session = new Session(); new_session.originate(session, "{ignore_early_media=true}sofia/gateway/sip.sipdiscount.com/0056979039388", 30); new_session.waitForAnswer(10000); // IF everybody is ready, then bridge our current session & the new_session if (session1.ready() && new_session.ready()) { console_log("info", "Interoligofrenico!\n"); bridge(session1, new_session); } // hangup when done session1.hangup(); new_session.hangup(); When I register with extension 1000 using a softphone and dial extension 500, the code above successfully creates the two new sessions, makes the calls in order, and supposedly bridges the sessions (I get no error about the bridge on the console, and I get the log text too), but I get no audio on either end. Anybody know what's wrong with the code? Btw, how can I originate a call to a configure extension? All examples on the wiki use the 'sofia syntax', should I just use sofia/default/1001 for example? Thanks! Nicolas

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Re: [Freeswitch-users] Freeswitch-users Digest, Vol 21, Issue 73
by Henk Oegema 28 Mar '08

28 Mar '08

> Message: 2 > Date: Thu, 27 Mar 2008 17:05:43 -0500 > From: Brian West <brian.west(a)mac.com> > Subject: Re: [Freeswitch-users] How to reply ? > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <CF21EBC2-F296-468E-BC9B-9CE1C3ED3B3F(a)mac.com> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > Press the reply button in your email program and they'll thread > automatically. > > /b > But I get all messages in a bunch, one or twice a day. So I can't reply to a single message. May be I need to change my settings? So let me see what happens to this message. (I deleted the rest of this Vol21, Issue 73)

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Getting Started Answering Calls
by Brian Tanner 28 Mar '08

28 Mar '08

Hi there. I might end up being active in this list, so I'll introduce myself. My name is Brian, I'm a complete newbie to telephony systems in general. I'm sorry, I know that can be frustrating, I don't really know the jargon and I don't really understand how things work. The good news is that I am good at figuring things out and I like to fiddle. Let me see if I have my jargon right: I have a DID mapped to AIX with a service provider (DID WorldWide). I have a cable modem at home, and my laptop (where FreeSwitch is) is the DMZ. I haven't touched the FreeSwitch config. When I call my local DID, FreeSwitch answers: freeswitch(a)bt-macBook.local> 2008-03-27 16:53:45 [NOTICE] mod_iax.c: 1011 mod_iax_runtime() Incoming call connected 17807010953, 7806692738, (null) 256/2097023 2008-03-27 16:53:45 [NOTICE] mod_iax.c:1019 mod_iax_runtime() New Inbound Channel (null)! 2008-03-27 16:53:45 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel IAX/17807010953-f30d [14617d63-9bb8-428b-8207-22a90693ae42] 2008-03-27 16:53:45 [INFO] mod_iax.c:399 iax_set_codec() Activate Codec PCMU/8000 20 ms 2008-03-27 16:53:45 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing ->17807010953@default 2008-03-27 16:53:45 [NOTICE] switch_ivr.c:1001 switch_ivr_session_transfer() Transfer IAX/17807010953-f30d to enum[17807010953@default] 2008-03-27 16:53:46 [INFO] switch_core_state_machine.c:113 switch_core_standard_on_ring() No Route, Aborting 2008-03-27 16:53:46 [NOTICE] switch_core_state_machine.c:114 switch_core_standard_on_ring() Hangup IAX/17807010953-f30d [CS_RING] [NO_ROUTE_DESTINATION] 2008-03-27 16:53:46 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 1 (IAX/17807010953-f30d) Ended 2008-03-27 16:53:46 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel IAX/17807010953-f30d [CS_HANGUP] Bus error (core dumped) So, the crash sucks, and doesn't happen every single time. I'm not tooo worried about that, because things are broken above. It looks like the call was answered (and it even knows who called it -- cool). But, it looks like FreeSwitch doesn't know what to do with the call, and so it aborts and hangs up. I think this is expected behavior. I want to use FreeSwitch for a few things, but the first is to build an IVR and Voicemail system. Can someone point me to resources or give me some quick tips on how to start this? There appear to be almost hundreds of config files, and it's all new to me, so I don't really know where to start, which files do what, etc, etc. If someone could help me with minimal instructions on how to play a sound file based on some control-flow code I write, or some way to run a script, or whatever, I'd be very grateful. I'm just really overwhelmed and I don't know how to start. Thanks in advance, very much! PS: If people help me get rolling, I'm happy to update wikis and/or write little tutorials as I go to save this hassle for future... -- Brian Tanner Ph.D Student, University of Alberta brian(a)tannerpages.com

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How to reply ?
by Henk Oegema 27 Mar '08

27 Mar '08

How do I reply to a message, so that my answer to a question remains in the same thread. (i.s.o. opening a new thread) ?

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SIP dialout problems
by Nicolas Brenner 27 Mar '08

27 Mar '08

Hi, I'm having trouble dialing out some phone numbers through a SIP gateway. I configured the gateway on conf/sip_profiles/outbound/gizmo.xml as: <include> <gateway name="gizmo"> <param name="debug" value="1"/> <param name="username" value="1747xxxxxx"/> <param name="from-domain" value="proxy01.sipphone.com"/> <param name="password" value="MYPASSWORD"/> <param name="caller-id-in-from" value="true"/> <param name="proxy" value="proxy01.sipphone.com"/> <param name="expire-seconds" value="3600"/> <param name="register" value="true"/> <param name="retry_seconds" value="3600"/> </gateway> </include> Then I created an extension on conf/dialplan/public.xml like this: <extension name="chile"> <condition field="destination_number" expression="^0056(.*)"> <action application="bridge" data="sofia/gateway/gizmo/0056$1"/> </condition> </extension> The above catches all calls that begin with the 0056 prefix (calls to Chile). So what I do is, register with a softphone using extension 1000, then I dial 0056xxxxx, and I get a recording saying the extension I'm calling is unavailable. The weird thing is that if I don't use 0056, the call is made, but then I get a recording from Gizmo (my test gateway) saying the number is wrong (of course). If instead of an expression, I dial the fixed number 1-800-466-4411 (Google 411), the called is successfully bridged. Also if I dial a Gizmo user number (1747xxxxxx) the call works. What's the problem with 0056? Apparently the call is never "leaving" freeswitch. For an unsuccessful call, I get this on the console: 2008-03-27 12:43:59 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1000@medularis01.dyndns.org:5060 [dbbaff68-0791-4253-a07c-ab2c4130df18] 2008-03-27 12:43:59 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing Nico->005624949458@default 2008-03-27 12:43:59 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/outbound/005624949458 [c0b1791c-eb96-464a-bc30-5d6481ed0c10] 2008-03-27 12:44:02 [NOTICE] sofia.c:1950 sofia_handle_sip_i_state() Hangup sofia/outbound/005624949458 [CS_HOLD] [NORMAL_UNSPECIFIED] 2008-03-27 12:44:02 [INFO] mod_dptools.c:1514 audio_bridge_function() Originate Failed. Cause: NORMAL_UNSPECIFIED 2008-03-27 12:44:02 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 16 (sofia/outbound/005624949458) Ended 2008-03-27 12:44:02 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/outbound/005624949458 [CS_HANGUP] 2008-03-27 12:44:02 [NOTICE] mod_dptools.c:1541 audio_bridge_function() Hangup sofia/default/1000@mydomain:5060 [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2008-03-27 12:44:02 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 15 (sofia/default/1000@mydomain:5060) Ended 2008-03-27 12:44:02 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/1000@mydomain:5060 [CS_HANGUP] What does NORMAL_UNSPECIFIED mean?! I found this page on the wiki: http://wiki.freeswitch.org/wiki/Hangup_causes but it's only a list of the codes. The page says "The codes are documented in src\switch_channel.c" but there's no documentation there at all, just the code. Any help would be greatly appreciated. Thanks in advance for your time and help. Regards, -- Nicolás Brenner

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directory with several xl files
by Ivan C Myrvold 27 Mar '08

27 Mar '08

I am not sure if this is a bug, or if this is how it should behave: If you in conf/directory have two files, say n1.xml and n2.xml, and two directories n1 and n2, and n1.xml includes files from n1 and n2.xml includes files from n2, then registrations are not working properly. Is this correct? Maybe it is supposed to only be one xml file in conf/ directory? Ivan

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Can't follow Getting Started Guide SIP profile instructions
by Erland Lewin 26 Mar '08

26 Mar '08

Hello, I just built Freeswitch from SVN, and am looking at the wiki Getting Started guide documentation: http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Default_FreeSWITCH_Co… Some things are unclear to me: "The sofia.conf.xml file includes other xml files to define multiple "profiles"" Looking at the "sofia.conf.xml" file generated by the default install, I see no inclusion of other xml files. "The default profiles are located in: $PREFIX/conf/sip_profiles/PROFILE_NAME.xml -- where PROFILE_NAME is the name of the profile." No sip_profiles directory was generated when I installed FreeSwitch, am I expected to create this myself, and add configuration file lines to include these files from sofia.conf.xml? sofia.conf.xml doesn't seem to reference a sip_profiles directory? I'm new to FreeSwitch, but it seems like the documentation and the default config files don't agree completely, which makes it hard for a noob to get this sorted out. Thanks for any help! /Erland

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SIP trunks with FS
by michael mendel 26 Mar '08

26 Mar '08

Hello I have a question , how I use SIP TRUNKS with FS. Regards, Michael Mendel

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How to start with the windows console demo?
by Ritesh Singh 26 Mar '08

26 Mar '08

Hi, I am very new to the FreeSwitch. Can someone tell me in detail how to start with the console demo of freeswitch on windows. It will be great if I can get some information on how to make test call. My aim is to make a demo in which 2 users talking to each other using freeswitch as a bridge. Also I would like to have all the call details. Any help regarding this is highly appreciated. Thanks and Regards Ritesh

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Answering machine detection
by Aloysius Thevarajah Lloyd 26 Mar '08

26 Mar '08

Hello , Thank you for the answer to my previous question. Is there a way to identify the Answering machine detection in FreeSwitch? Thank you Lloyd

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