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Trying to send SIP MESSAGE to device
by Paulo Emerique 10 Apr '10

10 Apr '10

Hi all, this is my first day of hands on experience with FreeSwitch, which I have been following for many months now. We deploy Asterisk boxes, and now that Freepbx runs with Freeswitch, we intend to "migrate" to it as our main platform. I'm really enjoying FS, congratulations! Now for the actual question: The issue is I bought a GSM Gateway, wich I'm trying to send SMS thru it ... and the device documentation says to send a "SIP MESSAGE" to it like: MESSAGE sip: 1003@192.168.1.126:5060;transport=udp SIP/2.0 From: <SIP 1000(a)192.168.1.100>;tag=5031 To:<SIP: 1003(a)192.168.1.126> Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008(a)192.168.1.100 CSeq: 3 MESSAGE Contact: <sip: 1000(a)192.168.1.100> Max-Forwards: 16 Date: Tue, 08 Apr 2010 06:36:37 GMT User-Agent: Paulo p-hint: usrloc applied Content-Type: text/plain Content-Length: 17 96913511 hi paulo GSM Gateway is at IP 192.168.1.126 ... so I have tried to acomplish it by 2 ways: 1) using mod event sockets: I telnet to server and after auth, issue: sendevent SEND_MESSAGE profile: sipinterface_1 content-length: 18 content-type: text/plain user: 1003 host: 192.168.1.126 96913511 Hello Baby and returns: Content-Type: command/reply Reply-Text: +OK accepted I'm watching SIP traffic (using "sofia profile sipinterface_1 siptrace on") and I see no SIP packet going. then I tried 2) chat sip|1030@192.168.1.132|1003@192.168.1.126|Test message which returns: Error! Message Not Sent 2010-04-10 10:46:34.555510 [ERR] sofia_presence.c:109 Chat proto [dp] from [1030(a)192.168.1.132] to [1003(a)192.168.1.126] Test message Invalid Profile 192.168.1.126 So I tought: well, need to configura a profile (wich I understand is created on freepbx "SIP Interfaces" ... so I did, but FS console shows: 2010-04-10 10:25:59.681316 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_3 To collect more info, I tried to make similar steps to send SIP MESSAGE to a Granstream HT-486, so I created a "SIP Interface" to it, but returns same error msg: 2010-04-10 10:25:59.868661 [ERR] sofia.c:1347 Error Creating SIP UA for profile: sipinterface_4 Thank for your help, Paulo Cesar Emerique

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UPDATE: G.729 Codec Licensing *IS* Available
by Michael Collins 08 Apr '10

08 Apr '10

Greetings all, The FreeSWITCH team would like to let everyone know that we do indeed sell g.729 licenses for $10 each. Use this link to initiate a purchase: http://www.freeswitch.org/node/235 Note: licenses are available only for Linux-based systems at this time. Please stay tuned for updates. The INSTALL.txt file has very detailed instructions: http://files.freeswitch.org/g729/INSTALL.txt Keep in mind that a single license includes one encoder and one decoder, that is, it can transcode both directions of a single phone call. If you have any other questions please email us here or join us in #freeswitch on irc.freenode.net. -Michael

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Another IRC day in #freeswitch!
by Michael Collins 08 Apr '10

08 Apr '10

Hey everyone, come join us in #freeswitch on irc.freenode.net! We are going for 250 users. (No sockpuppets, please.) If you don't have an IRC client then visit http://www.freeswitch.org and click the Join Us On IRC link to use the Web IRC applet. See you in channel! -MC (IRC: mercutioviz)

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streaming with mod_shout
by vip killa 08 Apr '10

08 Apr '10

I'm trying to figure out how to stream to a shoutcast server. There is no examples in the WIKI I've been able to stream to an icecast server using the command "conference name record shout://username:password@host/stream.mp3" my shoutcast server does not require a username, so what would the command be for an actual shoutcast server?

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FreeSWITCH Conference Call - Moises Silva From Sangoma Talks FreeTDM!
by Michael Collins 07 Apr '10

07 Apr '10

Hello everyone! Moises (moy) is going to be speaking today about FreeTDM. Please join the conference call at noon Eastern, 9am Pacific time: http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_07 Moises will start his presentation at about 12:15/9:15. Be sure to bring your questions about TDM and Sangoma! -Michael

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fedora 12 compiler error and workaround
by Douglas Hubler 06 Apr '10

06 Apr '10

Here is the complier error (sorry for pasting it in here, but you'll see why it's important in a second...) Creating mod_amr.so... /usr/lib/gcc/i686-redhat-linux/4.4.3/../../../crt1.o: In function `_start': (.text+0x18): undefined reference to `main' collect2: ld returned 1 exit status quiet_libtool: link: gcc -I/home/dhubler/work/freeswitch/src/include -I/home/dhubler/work/freeswitch/src/include -I/home/dhubler/work/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DAMR_PASSTHROUGH -o .libs/mod_amr.so -Wl,-x .libs/mod_amr.o -lm /home/dhubler/work/freeswitch/.libs/libfreeswitch.so -L/home/dhubler/work/freeswitch/libs/apr-util/xml/expat/lib /home/dhubler/work/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /home/dhubler/work/freeswitch/libs/apr/.libs/libapr-1.a -lpq -luuid -lrt -lcrypt -lpthread -lssl -lcrypto -ldl -lz -lncurses -Wl,-rpath -Wl,/usr/local/freeswitch/lib -Wl,-rpath -Wl,/usr/local/freeswitch/mod For some reason "-shared" is not on the link line Don't ask me how i figured this out, but If I change in file build/modmake.rules SOLINK = -shared -Xlinker -x to SOLINK = --shared -Xlinker -x Everything compiles and freeswitch runs. Something in the gnu tool chain wants to strip out any "-foo" parameter to gcc but adding an extra "-" it allows it through. So i didn't find the problem, but i did find a weird workaround.

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bicc
by Wasim Baig 05 Apr '10

05 Apr '10

folks: has anyone given thought to mod_bicc? how much effort is required? -- wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 | peace be upon you ...

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hooked functions are not called while using one single session
by Francisco Scaramanga 02 Apr '10

02 Apr '10

Hello, I am writing my own freeswitch module and have implemented 2 hooks for reading and writing audio (I did not implement the endpoint interface). My problem is that the channel_write_frame and channel_read_frame callback-functions are only called if I make an intercept with another session (switch_ivr_intercept_session), but my usecase has only one single Session for getting and putting audio data in my module. How can I achieve this? scaram /* on channel init I answer the session and set hooks */ static switch_status_t channel_on_init(switch_core_session_t *session) { switch_channel_t *channel = switch_core_session_get_channel(session); switch_channel_answer(channel); switch_channel_set_state(channel, CS_EXCHANGE_MEDIA); switch_set_flag(tech_pvt, TFLAG_IO); switch_core_event_hook_add_write_frame(session, &channel_write_frame); switch_core_event_hook_add_read_frame(session, &channel_read_frame); } static switch_status_t channel_write_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id) { /* is only called when I intercept 2 Sessions */ } static switch_status_t channel_read_frame(switch_core_session_t *session, switch_frame_t *frame, switch_io_flag_t flags, int stream_id) { /* is only called when I intercept 2 Sessions */ } _________________________________________________________________ http://redirect.gimas.net/?n=M1003xHM5Adressen2 Ein Postfach für Alles – bei Hotmail 5 E-Mail-Adressen online verwalten!

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Call Hold - moh direction
by Suneel Papineni 31 Mar '10

31 Mar '10

Hi, I am testing Call Hold command and got a doubt. Please clarify me if this is correct. I made a call from extension 1 to FSComm and call is answered. Now I want to put extension 1 on hold, so call hold command is issued to FSComm as "api uuid_hold <Unique-ID of call leg of extension1>" This has put call on hold and I expected music on hold to be played to extension 1, but moh is played to FSComm and not to the extension. Whereas if I change the command to "api uuid_hold <Unique-ID of call leg of FSComm>", then moh is played to the extension. Clarify me if my expectation is wrong and it is working correctly. I am testing with FS version "freeswitch-1.0.5-20100321-0400" on Windows system. Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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Moving FreeSWITCH forward (release planning)
by Andrew Thompson 28 Mar '10

28 Mar '10

Many of you will skim this and think "I want to see what bkw/anthm/etc have to say before I act." Don't. The reason 1.0.5 has not been released so far is enough people have not stepped up, took the initiative, and done something/anything productive. (If you're actively contributing to code/wiki/jiras, I'm probably not talking about you.) I had a brief, possibly anger inducing chat with anthm on IRC just about a week ago. I'm not angry, and I hope he's not angry. What I want is to help move FreeSWITCH forward. I realize that by pointing out things that need to be done, I volunteer myself to do them. I accept that responsibility, and we can deputize anyone else with a pulse who wants to own a piece of the process. I think everyone agrees that 1.0.4 is pretty stale, "archaic" I think I've heard. We're now 29 weeks since 1.0.4, the longest we've been without a new release since 1.0.0 was released. I've spent the week studying what release process information I could find for other open source projects. I've distilled that as best as I can into a temporary wiki page. It may be wrong. It may not be appropriate for FreeSWITCH. But, it's there. If it's horribly wrong, it can be fixed. It can be expanded. It can be formalized, agreed upon, and adopted. The packagers and developers that come along in the future can look at it and say 'OK, this is how they do releases.' They can know when to expect new releases, and who to contact for distribution specific issues. I submit for your review: http://wiki.freeswitch.org/wiki/User:Tobarja/Release_Process_Improvement -- Andrew Thompson

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