March 2007 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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mod_dingaling error
by Kieran O'Loughlin 08 Mar '07

08 Mar '07

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.

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how to setup dialplan..
by anish sheth 05 Mar '07

05 Mar '07

hi, I would like to test out following call scenario. sip softphone <-------> freeswitch <-------> googletalk client basically I want to make a call from sip softphone to say sip:1234@<freeswitch ip>. this call should land on freeswitch and freeswitch should act a as b2bua and setup a call to user2(a)gmail.com from freeswitch, dingaling module logs in to googletalk network using user1(a)gmail.com I have read the dialplan document on the web but am still not sure how to change diffferent configuration files... I am able to configure dingaling module config so that user1(a)gmail.com is logged into the googletalk network. also when I make a call from softphone to 1234@<freeswitch ip> I do see an INVITE in the freeswitch debugs... any help is appriciated. if there are more documents online that describe different configuration options, please point me to that.. Thanks, anish _________________________________________________________________ Dont miss your chance to WIN 10 hours of private jet travel from Microsoft® Office Live http://clk.atdmt.com/MRT/go/mcrssaub0540002499mrt/direct/01/

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Dialplan
by Jonas Gauffin 05 Mar '07

05 Mar '07

Hello again everyone =) Is it possible to configure destinations/actions depending on the domain that the user is calling from? (Having multiple companies in the same switch, where each company should be able to use their own destination/action for the same number). Are there any plans on making an dial plan module that contacts a TCP server to get what to do? (Instead of reading the XML files or using the LDAP directory). Best regards, Jonas

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Dialplan documentation
by Jonas Gauffin 05 Mar '07

05 Mar '07

Hello I've created a new page in the wiki called "dialplan documentation" where I have started to document what can be done in the dialplan. I've used the sourcecode (mod_dialplan_xml), the wiki and a diaplan as reference. It would be nice if a developer could read it and any fact errors, I hope that there aren't too many. http://wiki.freeswitch.org/wiki/Dialplan_documentation I'm missing tables in the wiki, using lists is not as nice as tables (check dokuwiki it both got tables and a more natural wiki format) Best regards, Jonas

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G.729
by Nikolay Kolev 04 Mar '07

04 Mar '07

Sometime ago there was info that it will be possible to buy G.729 licenses from the owners of the project. What's the status of it? Will this be possible soon or later? I've read posts on the web that some people are hesitant to use FreeSwitch (and are stuck with Asterisk) mainly because of G.729 issues. Nikolay

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snds directory tree is where?
by Spencer Nassar 04 Mar '07

04 Mar '07

I'm using a sample dialplan from the tarball distribution, but commands like "say" are failing because I don't have the snds/ directory. I downloaded http://files.freeswitch.org/freeswitch- snapshot.tar.gz day before yesterday, and just checked again a few minutes ago. How can I get the /snds/ directory tree? Thanks! Spencer ==== mod_dptools.c:134 phrase_function() Execute spell(650xxxxxxx) lang (null) switch_ivr.c:4745 switch_ivr_phrase_macro() No language specified - Using [en] switch_ivr.c:4862 switch_ivr_phrase_macro() Handle say:[650xxxxxxx] (en) mod_sndfile.c:145 sndfile_file_open() Error Opening File [/snds/ digits/6.wav] [System error : No such file or direc

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Technology using for SoftSwitch
by lanhdnvn 123 03 Mar '07

03 Mar '07

Dear! I am a student.I am treating about newtechnology using for SoftSwitch.If you have document about this problem.Can you help me,please?Or can you direct for me address to find.Thanks! --------------------------------- The fish are biting. Get more visitors on your site using Yahoo! Search Marketing.

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