- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Build system/RPM's
by Mike Murdock 21 Mar '07

21 Mar '07

Thanks for all the effort and quick turn around on fixes Mike, it seems to be building well on fedora core 5, 6 and red hat EL. Michael Murdock VP Operations & Development Switchmaxx Product Line 125 N. Market. Suite 1520 Wichita KS. 67202 (316) 932-2693 -----Original Message----- From: Michael Jerris [mailto:mike@jerris.com] Sent: Tuesday, March 20, 2007 10:53 PM Eastern Standard Time To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Build system/RPM's Big updates today on this. Problems building on OSx appear to now be resolved. Problems building on debian and ubuntu also appear to be resolved. Please note, on the version of libtool. I have seen many systems that have 1.5.6 installed. I know it is a pain to upgrade, but 1.5.6 is badly broken. I currently have the bootstrap failing at the start with any libtool 1.5.6 or older, however I have not tried any versions between 1.5.6 and 1.5.22. Please note that the changes over the last couple days have been quite invasive. If you have any issues bootstrapping or building, please start by trying to checkout a fresh tree to a different directory and seeing if that will build. Please report back any additional errors as before and thanks for sticking with us through all of this. Mike > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch- > users-bounces(a)lists.freeswitch.org] On Behalf Of Michael Jerris > Sent: Tuesday, March 20, 2007 1:45 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: Re: [Freeswitch-users] Build system/RPM's > > This appears to be a symptom of libtool 1.5.6 which is quite broken, and > appears to be on many machines. Did you have 1.5.6 on previously? If > you ./libtool --version in the root FreeSWITCH(tm) source dir, what does > it give you? I just had this same thing happen on a box even after > upgrading to 1.5.22 because I installed libtool from source using the > default prefix /usr/local and the one in /usr was still taking > precedence. I had to install to /usr to get it to work right. > > Also, as a note, I changed the checks in our bootstrap as of svn r4683 > to no longer bootstrap with libtool 1.5.6 due to this issue. And > clearly we will need to have pre-bootstrapped source tarballs out there > very soon for those who do not have appropriate autotools versions. I > am working on that now. > > Mike _______________________________________________ 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

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Build system/RPM's
by Michael Jerris 21 Mar '07

21 Mar '07

Some updates on the build system below. Please read on! The bulk of the invasive changes to the FreeSWITCH(tm) autotools build system are now in tree. We will be continuing to refine the system over time, but the bulk of the changes are now in tree. The important changes are as follows: The build targets are now more in line with typical packages. make: will build all the sourcecode and libraries make install: will install. make uninstall: will uninstall. make clean: will clean the build files The old targets (installall, modules, sure, modclean, modwipe, megaclean, ect) still all generally behave as they did before. For the libraries we have in tree, we, for the most part, now statically link to them and do not install them. This should resolve much of the issues reported in regards to the need for LD_LIBRARY_PATH and system library conflicts. The FreeSWITCH(tm) build will now use a system version of the curl library if it detects a usable version at configure time. We will probably add some more libraries to this list over time, as we review any patches that we have in the libs that we have in tree, and get those patches back to the library maintainers. This will be a long process due to the amount of libraries and the amount of changes in many of them. We will likely be doing some changes to the url's you checkout from svn to give you options on what libs you download. We will continue to use statically linked libraries in certain cases where we see performance or other benefits to doing so. These will most likely be configure options. We have started the process of building binary packages for FreeSWITCH(tm) svn snapshots. The first of these are at: http://software.opensuse.org/download/server:/telephony/ We are looking for people who are willing to write the appropriate files for operating systems and distros that are not covered by that list. In particular, for BSD ports, gentoo, and debian. If you are willing to lend a hand, please drop me a line. All of the auto-tools generated files have now been removed from the svn tree. If you are building from svn, you will need to have autoconf, automake and libtool installed, and run ./bootstrap.sh and configure. I highly suggest using -C on your configure line, it speeds up the configure quite a bit. We will have a location for pre-bootstrapped source tarballs soon. In the mean time, if anyone could give me feedback on any boostrap/configure/make failures on different platforms. Known issues currently are as follows: OsX builds are currently broken unless you manually change apr to build shared libraries, and install them. This I plan on fixing today. Bsd users can now use the standard bsd make for everything except the sofia-sip library. I am also working on addressing this issue. Any other issues please let me know (best to track bugs on jira.freeswitch.org) and I will follow up with another update to the list soon. Mike.

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Compile problem
by Kieran O'Loughlin 20 Mar '07

20 Mar '07

Hey guys, I had an older version of freeswitch installed and running (from about 2 weeks ago). I'm having problems bridging a gtalk call to remote sip (local registration sip to remote sip works, remote sip to gtalk works but no audio on gtalk to sip). I downloaded the latest version using svn. When I tried to run "configure" it wasn't there. The only thing I could find was bootstrap.sh. I ran it but get a lot of errors like: automake: tests/server/Makefile.am: not supported: source file `$(top_srcdir)/lib/timeval.c' is in subdirectory utils/Makefile.am:46: variable `GLIB_LIBS' not defined Makefile.am:8: invalid unused variable name: `AM_LDFLAGS' If I keep going and and run the configure file created I get these errors: ./configure: line 1393: syntax error near unexpected token `lib/config.h' ./configure: line 1393: `AM_CONFIG_HEADER(lib/config.h src/config.h)' configure: error: /bin/sh './configure.gnu' failed for libs/curl Can anyone help? Thanks, Kieran.

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Freeswitch?
by Robert La Ferla 20 Mar '07

20 Mar '07

I am an Asterisk user but I hate the monolithic design. I would like to know more about the Freeswitch design and roadmap. I know that it is basically a lower level yet modular system. I have read the FAQ and wiki. I basically want to know what will it take to put together a Freeswitch system that: 1. Routes calls from my SIP extensions to either a SIP service provider (like Broadvoice) AND an PSTN provider (like Veriizon) 2. Has a voicemail system 3. Has dial and Caller ID rules for dialing (dial plan), answering, etc... Are these components in place yet? If not, will they be soon? Are there other open source packages that interoperate w/Freeswitch to accomplish the above?

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Re: [Freeswitch-users] FreeSwitch and Click 2 Call
by Anthony Minessale 19 Mar '07

19 Mar '07

update to 4657 and come into irc 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: Helmut Kuper <helmut.kuper(a)ewetel.de> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, March 19, 2007 8:34:12 AM Subject: Re: [Freeswitch-users] FreeSwitch and Click 2 Call -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Anthony, latest freeswitch trunks is much better. It sends now an OK message with a SDP. Unfortunately the contact Addres in SDP is empty (NULL), so that WeSIP resends INVITES. After some time freeswitch responses with "482, Request Merged" At jira u can find that sip flow. I'm right now on irc as G-Man regards Helmut Anthony Minessale schrieb: > Please update to the lastest svn trunk code > and tell me what happens. I took a wild guess a how > to handle this because I had no test setup. > > > If it still does not work you will need to > communicate with us in realtime and setup a test > for use to work on to implement your feature. > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF/pFU4tZeNddg3dwRAho3AJ9ri5g99NJufDLxCdRrDIO1+L9MuQCcDqRF y983h8D1SwyM8/vpG34fDi8= =lR7F -----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 ____________________________________________________________________________________ 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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Portaudio Compile error
by Andrey Kropachev 19 Mar '07

19 Mar '07

Hello, I'm trying to compile Freeswitch on a Freebsd 6.1 machine and get the following error: ------------------------------------------------------------------------------ making endpoints/mod_portaudio gcc -c -g -O2 -DPA_LITTLE_ENDIAN -pthread -I./include -I./src/common -I./src/os/unix -DPACKAGE_NAME=\"\" -DPACKAGE_TARNAME=\"\" -DPACKAGE_VERSION=\"\" -DPACKAGE_STRING=\"\" -DPACKAGE_BUGREPORT=\"\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 -DHAVE_DLFCN_H=1 -DSIZEOF_SHORT=2 -DSIZEOF_INT=4 -DSIZEOF_LONG=4 -DHAVE_CLOCK_GETTIME=1 -DHAVE_NANOSLEEP=1 -DPA_USE_OSS=1 src/common/pa_ringbuffer.c -fPIC -DPIC -o src/common/pa_ringbuffer.o src/common/pa_ringbuffer.c:75:33: libkern/OSAtomic.h: No such file or directory *** Error code 1 Stop in /usr/home/spartan/freeswitch/freeswitch_snapshot/libs/portaudio. *** Error code 1 ------------------------------------------------------------------------------ How can I fix this problem& Best regards, Andrey Kropachev mailto:a.kropachev@vktv.ru

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Re: [Freeswitch-users] FreeSwitch and Click 2 Call
by Anthony Minessale 19 Mar '07

19 Mar '07

Please update to the lastest svn trunk code and tell me what happens. I took a wild guess a how to handle this because I had no test setup. If it still does not work you will need to communicate with us in realtime and setup a test for use to work on to implement your feature. 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: Helmut Kuper <helmut.kuper(a)ewetel.de> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, March 16, 2007 4:50:42 AM Subject: Re: [Freeswitch-users] FreeSwitch and Click 2 Call -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, to get this problem with freeswitch, I use following environment: - -latest freeswitch source - -openser 1.2.0 as wesip frontend - -wesip 0.1.3 as 3pcc application all on one machine I register as 101(a)sip0.en.ewetel.de at freeswitch I enter in wesip c2c web application 101(a)sip0.en.ewetel.de as caller and a PSTN phone as callee. The click dial button. WeSIP sends through openser an INVITE without SDP to freeswitch targeting 101(a)sip0.en.ewetel.de. freeswitch replys with Trying and comes never back. We had to expect here that my sip phone (101@...) rings but nothing happens. When I accept that call an Ok with sdp offer shall be generated and send through freeswitch back to openser and wesip. regards Helmut Anthony Minessale schrieb: > What are you using to trigger this? > I can't really support it without a way to try it. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF+mhy4tZeNddg3dwRAv/dAKCneqX/mNMLa/r2F7v/mGK0Tgj5wQCgns2J vTDk6McftjugnOLJigB5F+A= =+3Hb -----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 ____________________________________________________________________________________ Don't get soaked. Take a quick peek at the forecast with the Yahoo! Search weather shortcut. http://tools.search.yahoo.com/shortcuts/#loc_weather

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Conference and sound file questions
by Jonas Gauffin 17 Mar '07

17 Mar '07

Hello I would like to get more clear examples on how to configure the conference module. First of all, the documentation states that the dialplan action should look like this: <action application="conference" data="confname@profile+flags{mute|deaf}+[conference pin number]"> Would a real world example look like this: <action application="conference" data="myconf@default+flags{mute}+[1234]" /> or should the square brackes be removed? Next, what does "confname" and "profile" correspond to when looking in the conference config? I can understand that "profile" is the same as in <profile name="xxxx">, but what are the confname? (do confname stand for configuration name, or conference name?) ---------- What kind of sound file formats are supported? And what kind of format should the formats be in? =) For instance if wave files are supported, should they be 8khz 8bit mono or something else? //Jonas (i'll add the information to the wiki too)

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Re: [Freeswitch-users] FreeSwitch and Click 2 Call
by Anthony Minessale 16 Mar '07

16 Mar '07

What are you using to trigger this? I can't really support it without a way to try it. 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: Helmut Kuper <helmut.kuper(a)ewetel.de> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, March 15, 2007 11:50:56 AM Subject: Re: [Freeswitch-users] FreeSwitch and Click 2 Call -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, is already there ... regards Helmut Brian West schrieb: > Please attach it to the bug report in jira! > > /b > > On Mar 15, 2007, at 7:26 AM, Helmut Kuper wrote: > > Hi, > > > please find attached a working 3pcc sip flow trace > > > regards > helmut > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF+Xlu4tZeNddg3dwRAsI2AKCs5a2KZftobnKlbFq2DjyVCIsdygCcChXQ jmacrzmRX1LKip4tjma13WM= =i1/q -----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 ____________________________________________________________________________________ Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains. http://farechase.yahoo.com/promo-generic-14795097

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Re: [Freeswitch-users] FreeSwitch and Click 2 Call
by Anthony Minessale 15 Mar '07

15 Mar '07

I did my best to deal with it with no way to test it. 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: Helmut Kuper <helmut.kuper(a)ewetel.de> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, March 15, 2007 11:50:56 AM Subject: Re: [Freeswitch-users] FreeSwitch and Click 2 Call -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, is already there ... regards Helmut Brian West schrieb: > Please attach it to the bug report in jira! > > /b > > On Mar 15, 2007, at 7:26 AM, Helmut Kuper wrote: > > Hi, > > > please find attached a working 3pcc sip flow trace > > > regards > helmut > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF+Xlu4tZeNddg3dwRAsI2AKCs5a2KZftobnKlbFq2DjyVCIsdygCcChXQ jmacrzmRX1LKip4tjma13WM= =i1/q -----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 ____________________________________________________________________________________ Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html

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