- FreeSWITCH-users - FreeSWITCH Mailing Lists

User Agents and extensions
by Jonas Gauffin 15 Mar '07

15 Mar '07

Hello When I configure user agents I setup a name and a password, right? How do I map a UA to a specific extension? ---- I have to UA's configured and both have logged in sucessfully (registered in freeswitch). I tried to dial jonas(a)pbx.gauffin.org from micke(a)pbx.gauffin.org, but I got a 404 back. Do I need to configure the dialplan to get that case working or have I missed something else? //Jonas

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FreeSwitch and Click 2 Call .. Ala SNAPNumber
by EdPimentl 14 Mar '07

14 Mar '07

What would be the best way to implement http://SnapNumber.com in FS? Here is the description of SNAP Snap is a cutting edge dialer and call pop up application for Asterisk that is flexible and powerful. Snap works by sending the phone number you wish to dial to your PBX and initiating a call back to your phone. Once your phone rings you pick it up and it will be connected to the number you dialed. For Outlook users, the call pop-ups are tightly integrated with your contacts and will show their picture (if available) and display name. The flexibility comes from it's Multi-Connection technology. If you travel between work and home, or would like to have seperate settings for different situations then this feature will be extremely useful for you. You simply dial via a different connection by using the "arrow" to the right of the Dial button to access these connections. Snap integrates with Outlook, Excel, Powerpoint, and Thunderbird. The *Firefox and Thunderbird plugin's* automatically detect phone numbers on webpages and make them dialable by simply clicking -E Thanks in advance and best regards, Ed Mail: edpimentl[at]gmail.com sip: sip:2150112@truphone.com IM: edpimentl [AOL | Jabber | Yahoo | MSN ] Voip: edpimentl [SKype | GoogleTalk ] Mobile Content Marketing/Management/Digital Delivery http://mobilecentral.ws Mobile ( Relevant, Ambient, Location ) based Social Network http://TagR.mobi (Alpha) Private Label Social Networks http://6bacon.com http://GooGaYa.com Mobile Payment - P2P Payment http://agilepay.ws Sponsor of P2PSIP open source [viasip_ng] project Creating a standards base P2PSIP dSIP infrastructure https://sourceforge.net/projects/viasip/ Mailing List viasip_ng(a)freelists.org

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FreeSwitch and Click 2 Call
by Helmut Kuper 13 Mar '07

13 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I downloaded today the actual SVN trunk (4488) of freeswitch and compiled it. I tested again my C2C szenario and have to say it still doesn't work. I get this console output of freeswitch: tport_recv_iovec(0x80a3780) msg 0x80afaa0 from (udp/212.6.123.118:5090) has 513 bytes, veclen = 1 recv 513 bytes from udp/[212.6.123.118]:5060 at 12:34:53.157657: ------------------------------------------------------------------------ INVITE sip:101@212.6.123.118:5090 SIP/2.0 Record-Route: <sip:212.6.123.118;lr=on;ftag=8BCAB768E8A4A50A49EC948C2C75711E> Via: SIP/2.0/UDP 212.6.123.118;branch=z9hG4bK763e.f253e907.0 Via: SIP/2.0/UDP 212.6.123.118:5080;branch=z9hG4bK763e.847316c5.0 To: sip:101@sip0.en.ewetel.de From: sip:101@sip0.en.ewetel.de;tag=8BCAB768E8A4A50A49EC948C2C75711E CSeq: 1 INVITE Call-ID: 2ECB15547F3AF88ECA67FF1C37AF5694(a)212.6.123.118 Content-Length: 0 Max-Forwards: 69 Contact: <sip:212.6.123.118:5080;transport=udp> ------------------------------------------------------------------------ tport_deliver(0x80a3780): msg 0x80afaa0 (513 bytes) from udp/212.6.123.118:5090/sip next=(nil) nta: received INVITE sip:101@212.6.123.118:5090 SIP/2.0 (CSeq 1) nta: canonizing sip:101@212.6.123.118:5090 with contact nta: INVITE (1) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x80a1e10, 0x80a0030, 0x8099840) called soa_set_params(static::0x80a8760, ...) called soa_set_params(static::0x80a8760, ...) called nta_leg_create(0x80a86f8) soa_init_offer_answer(static::0x80a8760) called nua(0x8099840): adding session usage tport_tsend(0x80a3780) tpn = UDP/212.6.123.118:5060 tport_resolve addrinfo = 212.6.123.118:5060 tport_by_addrinfo(0x80a3780): not found by name UDP/212.6.123.118:5060 tport_vsend(0x80a3780): 458 bytes of 458 to UDP/212.6.123.118:5060 tport_vsend returned 458 send 458 bytes to udp/[212.6.123.118]:5060 at 12:34:53.158826: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.6.123.118;branch=z9hG4bK763e.f253e907.0 Via: SIP/2.0/UDP 212.6.123.118:5080;branch=z9hG4bK763e.847316c5.0 Record-Route: <sip:212.6.123.118;lr=on;ftag=8BCAB768E8A4A50A49EC948C2C75711E> From: sip:101@sip0.en.ewetel.de;tag=8BCAB768E8A4A50A49EC948C2C75711E To: sip:101@sip0.en.ewetel.de Call-ID: 2ECB15547F3AF88ECA67FF1C37AF5694(a)212.6.123.118 CSeq: 1 INVITE User-Agent: FreeSWITCH(mod_sofia) Content-Length: 0 ------------------------------------------------------------------------ nta: sent 100 Trying for INVITE (1) nua: nua_stack_event: entering nua(0x8099840): i_invite 100 Trying nua(0x8099840): call state changed: init -> received nua: nua_stack_event: entering nua(0x8099840): i_state 100 Trying nua: nua_event: entering nua: nua_handle_bind: entering nua: nua_event: entering 2007-03-09 13:34:53 [NOTICE] switch_channel.c:347 switch_channel_set_name() New Chan sofia/sip/101@sip0.en.ewetel.de:5060 [9473b37c-ce3a-11db-bcb2-5d58d266c510] 2007-03-09 13:34:53 [DEBUG] mod_sofia.c:2895 sip_i_state() Channel sofia/sip/101@sip0.en.ewetel.de:5060 entering state [received] tport_wakeup_pri(0x80a3780): events IN tport_recv_event(0x80a3780) tport_recv_iovec(0x80a3780) msg 0x80ad9c0 from (udp/212.6.123.118:5090) has 4 bytes, veclen = 1 tport_deliver(0x80a3780): bad msg 0x80ad9c0 (4 bytes) from udp/212.6.95.198:5090/sip next=(nil) nta_agent: received garbage from udp/212.6.95.198:5090/sip after that there is no more action on this INVITE. Shutdown of freeSwitch brings this up: nta_agent: received garbage from udp/212.6.95.198:5090/sip 2007-03-09 13:48:16 [CONSOLE] switch_core.c:4621 switch_core_destroy() End existing sessions 2007-03-09 13:48:16 [NOTICE] switch_core.c:713 switch_core_session_hupall() Hangup sofia/sip/101@sip0.en.ewetel.de:5060 [CS_NEW] [SYSTEM_SHUTDOWN] 2007-03-09 13:48:16 [INFO] switch_channel.c:992 switch_channel_perform_hangup() Kill sofia/sip/101@sip0.en.ewetel.de:5060 [KILL] 2007-03-09 13:48:16 [INFO] switch_core.c:714 switch_core_session_hupall() Kill sofia/sip/101@sip0.en.ewetel.de:5060 [KILL] tport_wakeup_pri(0x80a3780): events IN tport_recv_event(0x80a3780) tport_recv_iovec(0x80a3780) msg 0x80ad9c0 from (udp/212.6.123.118:5090) has 4 bytes, veclen = 1 tport_deliver(0x80a3780): bad msg 0x80ad9c0 (4 bytes) from udp/212.6.95.198:5090/sip next=(nil) nta_agent: received garbage from udp/212.6.95.198:5090/sip 2007-03-09 13:48:27 [CONSOLE] switch_core.c:4623 switch_core_destroy() Clean up modules. regards helmut - -- Mit freundlichen Grüßen Helmut Kuper Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.kuper@ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Straße 310 26133 Oldenburg Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Geschäftsführung: Hans-Joachim Iken, Dirk Thole, Jürgen Wehlend Homepage: http://www.ewetel.de ___________________________________ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF8Vg04tZeNddg3dwRAlpCAJ94/eNqv/XlC5vNC0u+CczSgQM8dwCfbqwi hufthKDv3zOrG9VjcC4GtqE= =TUyY -----END PGP SIGNATURE-----

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build system changes.
by Michael Jerris 10 Mar '07

10 Mar '07

Hot off the svn commit, there are now some changes in tree that are the first steps towards our new auto-tools build system. The primary change is that we now enforce usage of the switch_ namespace for apr, apr-utils, pcre, and sqlite. There is also some fixes as to how we setup our include paths for header files, that should help fix some of the issues we have had with header files that were included not being the ones from your current source tree. PLEASE NOTE: if you are building using an auto-tools build (everyone except msvc builds currently) you will NEED to do a "make megaclean" and a new configure after you update to this new code. If you encounter any issues please report them as soon as possible on our bug-tracker at http://jira.freeswitch.org . More changes to the build system will continue to flow into tree over the next week, so if you run into issues, please first try a "make megaclean" and new configure before reporting a problem. Thanks Mike

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Click2Call question
by Helmut Kuper 09 Mar '07

09 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to use WeSIPs Click2Call application with freeswitch. freeswitch gets the invite (with no SDP) from WeSIP according to http://www.wesip.eu/mediawiki/index.php/Applications:Click2Dial-_Signalling or RFC 3725 Chapter 4.1 (http://www.ietf.org/rfc/rfc3725.txt) Unfortunately freeswitch never sends back the required OK message as described in RFC 3725 Chapter 4.1 Is there any patch for it or is RFC 3725 support planned for freeswitch? regrads helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF7shK4tZeNddg3dwRAsosAJ9rE9bfIWB/anp5e/ZLBTUbLd4R+QCcC7Y1 oLV2RNtJnNsd6Y9cDBwFKvY= =kJtq -----END PGP SIGNATURE-----

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Re: [Freeswitch-users] mod_dingaling error
by Anthony Minessale 08 Mar '07

08 Mar '07

The iksemel library that we use in libdingaling needs to have GNUtls pre-installed to work properly Follow these steps for unix 1) Install the gnutls package for your OS. 2) go to the freeswitch build root 3) execute..$rm -f libs/iksemel/.complete 4) execute..$make sure For windows see the other thread going around today. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Kieran O'Loughlin <kieran(a)alumni.virginia.edu> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, March 8, 2007 5:09:13 AM Subject: [Freeswitch-users] mod_dingaling error Hey all, I've been trying to get the gtalk connectivity working for a while now. I've been having problems that I think are related to NAT issues. The signaling piece of the call was working, but the media piece wouldn't go through. In any case I decided to try the install on a friend's machine that has a public IP and is not behind NAT. His is a debian box with a 2.4 kernel. I used exactly the same dingaling.conf.xml file as on my own installation (which is registering just fine on my box). When I start up freeswitch it's not registering with gtalk. There error appears to be related to TLS not being supported. However I have exactly the same version of freeswitch running on my own box and it uses TSL quite happily. Here is the error I'm getting: freeswitch@spock> SEND[<?xml version='1.0'?><stream:stream xmlns:stream='http://etherx.jabber.org/streams ' xmlns='jabber:client' to='gmail.com' version='1.0'>] RECV[<?xml version="1.0" encoding="UTF-8"?><stream:stream from=" gmail.com" id="CEE35FAE16C190D5" version="1.0" xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client"><stream:features><starttls xmlns="urn:ietf:params:xml:ns:xmpp-tls"><required/></starttls><mechanisms xmlns="urn:ietf:params:xml:ns:xmpp-sasl"><mechanism>X-GOOGLE-TOKEN</mechanism></mechanisms></stream:features>] 2007-03-08 12:18:16 [DEBUG] libdingaling.c:1096 on_stream() TLS NOT SUPPORTED IN THIS BUILD! Here is the my dingaling.conf.xml file: <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="1"/> <param name="codec-prefs" value="PCMU@8000"/> </settings> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="me(a)gmail.com/talk"/> <param name="password" value="password"/> <param name="dialplan" value="XML"/> <param name="message" value="Jingle-to-SIP Gateway via Freeswitch"/> <param name="auto-login" value="true"/> <param name="auto-reply" value="Press *Call* to call me via FreeSWITCH"/>  <param name="sasl" value="md5"/>  <param name="server" value="talk.google.com:5222 "/>  <param name="tls" value="true"/>  <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto"/>   <param name="exten" value="1004j"/>    <param name="vad" value="both"/> </profile> </configuration> I suspect it's because there are some libraries that are missing on the debian box when I compiled it. I installed libgnutls-dev and recompilted freeswitch but no joy. Thanks, Kieran. _______________________________________________ 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 ____________________________________________________________________________________ The fish are biting. Get more visitors on your site using Yahoo! Search Marketing. http://searchmarketing.yahoo.com/arp/sponsoredsearch_v2.php

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(GNU)TLS and VC++ Edition
by Peter Reimer 08 Mar '07

08 Mar '07

Hi, I'm trying to get freeswitch work together with google talk. Freeswitch has been compiled with Visual C++ Express Edition without any changes in the sources but I get the message "TLS isn't supported in this build". I read somewhere that I have to install GNUTLS and then I did so and added the include- and lib path of gnutls to the project - without any success. Then I looked around in the sources and found: #ifdef HAVE_GNUTLS ... so I added /D "HAVE_GNUTLS" to the iksemel properties and tried to rebuild iksemel but I got many errors (syntax errors in gnutls.h). This seems to be the wrong way... Does anyone have an idea? Greetings, Peter ___________________________________________________________ Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de

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Re: [Freeswitch-users] Call on Demand
by Anthony Minessale 08 Mar '07

08 Mar '07

Look at mod_event_socket or mod_xml_rpc and the "originate api command" as well as the bridge conference and possibly mod_spidermonkey. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: EdPimentl <edpimentl(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, March 8, 2007 7:28:42 AM Subject: [Freeswitch-users] Call on Demand Hello Fellow FS, How would this be done in FS Want to be able to have FS accept a call to action " CALL | CONF " take one or more SIP URI or e164 and send and invite to it. .CALL 6785228771 7703324512 Support Accepting SIP URI(sip:edpimentl@agilesip.com one or more), e164 number(ie 6785331111 one or more) and establish a call .CALL. 678522877 14:00 EASTERN .SAY. Will Not be available for lunch Support dialing out a number at certain time and delivered a Txt2speech voice message. .CONF. 6784441231 7703331212 PIN413121# Support Conference - Create Adhoc conf call with preassigned pin PIN Thanks in advance, -E _______________________________________________ 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 ____________________________________________________________________________________ Never miss an email again! Yahoo! Toolbar alerts you the instant new Mail arrives. http://tools.search.yahoo.com/toolbar/features/mail/

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Re: [Freeswitch-users] Click2Call question
by Anthony Minessale 08 Mar '07

08 Mar '07

We already have a stub mod_pjsip.c with promise of a GPL exception for non-commercial use and licensing options available for commercial use. I also have a mod_sipx.cpp lying around somewhere and of corse good ol' mod_exosip. If you want to play with any of them let me know. FreeSWITCH is designed to have mutiple endpoints so as long as they do not try to use the same IP/Port they can co-mingle. Be aware that we have tried many solutions and sofia to date has been able to provide the most reliability with pjsip comming in as a 2nd place hopeful but seeming to fail on some race conditions and not guaranteed to work right on 64bit. Perhaps pick some less-developed area of the project to enhance =D Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: "ed(a)edpimentel.com" <ed(a)edpimentel.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, March 8, 2007 6:48:40 AM Subject: Re: [Freeswitch-users] Click2Call question Is it possible to use plug in another SIP-STACK ie http://www.pjsip.org/pjsip/docs/html/index.htm Or you can also dovetail FS with http://www.jivesoftware.com/products/wildfire/featuretour.jsp And use Smack API and XPP http://www.igniterealtime.org/ http://www.extreme.indiana.edu/xgws/xsoap/xpp/ The Holy Grail would be when we can plug in Jay's AdHEARsion engine http://adhearsion.com/ to FS To that point, let me know who would want to sponsor with me a port of AdHEARsion to FS with me. Best, -Edpimentl > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi, > >> You have to remember that FreeSWITCH is not a sip stack... Sofia-SIP >> is. We will need to raise this issue with them. > > hm you are right ... :) > > >>Why is it sending an invite without an SDP? > > Well dunno. I guess it's because a webapplication starts the INVITES to > both caller and callee and for the first one to Caller the application > has no idea about the SDP data for caller. the SDP is send in a 200 OK > message from caller to freeswitch and from there back to web > application. After that the web app invites the calle with sdp from > caller ... > > regards > helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.5 (MingW32) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFF7tW34tZeNddg3dwRAmaXAJ9RWyGBVw/O0hB0ai8qi/zT82yDRQCfYleI > isydW+Wgdl1hed44SZOqAy4= > =ZUel > -----END PGP SIGNATURE----- > > _______________________________________________ > 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 > _______________________________________________ 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 ____________________________________________________________________________________ The fish are biting. Get more visitors on your site using Yahoo! Search Marketing. http://searchmarketing.yahoo.com/arp/sponsoredsearch_v2.php

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Call on Demand
by EdPimentl 08 Mar '07

08 Mar '07

Hello Fellow FS, How would this be done in FS Want to be able to have FS accept a call to action " CALL | CONF " take one or more SIP URI or e164 and send and invite to it. .CALL 6785228771 7703324512 Support Accepting SIP URI(sip:edpimentl@agilesip.com one or more), e164 number(ie 6785331111 one or more) and establish a call .CALL. 678522877 14:00 EASTERN .SAY. Will Not be available for lunch Support dialing out a number at certain time and delivered a Txt2speech voice message. .CONF. 6784441231 7703331212 PIN413121# Support Conference - Create Adhoc conf call with preassigned pin PIN Thanks in advance, -E

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