March 2007 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] G722 8000 vs 16000
by Anthony Minessale 21 Mar '07

21 Mar '07

I fixed it the best I could (At least for SIP) The stupid thing here is that there really is an 8khz g722 and the moronic lemmings who would implement the *wideband* version to say 8000 despite the fact that they *know* it is 16000 and are too stupid to realize should be the ones to suffer not us. sigh, I moved actual 8k g722 to payload 109 name G722_8 and put 16k g722 on payload 9 by itself and added a hack to the sdp generator in mod_sofia to display 8000 This makes my stomach turn but I guess we must do whatever it takes to preserve stupidity. This makes me wonder if more of the SIP and SDP RFC's are just typo's which would explain a lot.... 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: "Sluschny, Thomas" <Thomas.Sluschny(a)siemens.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 20, 2007 11:42:17 AM Subject: [Freeswitch-users] G722 8000 vs 16000 Hi, i want to use G.722 with its wide audio bandwith of 7kHz. But i have found that FreeSwitch resamples always from 16000 to 8000 Samples/s if i connect a "standard device", i. e. a SIP device with a "G722:9/8000" in its SDP. On the other side it is possible to use full bandwith if there is a "G722:9/16000" in SDP. I think this is not correct! FreeSwitch should always use 16000S/s and also use "G722:9/8000" in SDP! Read RFC3551, 4.5.2: ...Even though the actual sampling rate for G.722 audio is 16,000 Hz, the RTP clock rate for the G722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged for backward compatibility. ... I tried the 16kHz-implementation with a SIP phone (siemens optipoint 410 with "G722:9/8000") and it seems to work. In the g722.h file is mentioned that the 8k option is used for fast and flexible interworking with narrow band telephony. I think this is a special case (we also support G711 for this). Our default should be 16000S/s and "G722:9/8000". What do u think? Thomas _______________________________________________ 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 ____________________________________________________________________________________ Need Mail bonding? Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. http://answers.yahoo.com/dir/?link=list&sid=396546091

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Bridging dingaling to sofia
by Kieran O'Loughlin 21 Mar '07

21 Mar '07

Hey all, I've been playing around with trying to bridge dingaling to sofia for some time now. I've had some success. 1) I can call to googletalk using a local sip registration 2) I can call to googletalk using a remote sip registration 3) I can call from googletalk to a local sip registration 4) I can call from a local sip registration to a remote sip registration In case my terminology is bad when I say local registration I mean that the soft-phone is registered with freeswitch. When I say remote sip registration I mean that the soft-phone is registered with my standard SIP provider. The problem is if I call from googletalk and attempt to bridge the call to my remote sip provider the call rings, but there is no audio. I've tracked through the console and here is the piece that's different. This piece never shows up if I attempt to bridge the call to a remote sip registration. I copied this from the console when bridging to a local sip registration. 2007-03-21 12:56:13 [DEBUG] mod_sofia.c:4982 event_callback() event [nua_r_invite] status [180][Ringing] session: sofia/XX.XX.XX.XX/kieran 2007-03-21 12:56:13 [DEBUG] mod_sofia.c:4982 event_callback() event [nua_i_state] status [180][Ringing] session: sofia/XX.XX.XX.XX/kieran 2007-03-21 12:56:13 [DEBUG] mod_sofia.c:2991 sip_i_state() Channel sofia/XX.XX.XX.XX/kieran entering state [proceeding] 2007-03-21 12:56:13 [NOTICE] mod_sofia.c:3018 sip_i_state() Ring-Ready sofia/XX.XX.XX.XX/kieran! 2007-03-21 12:56:13 [INFO] switch_core.c:1872 2007-03-21 12:56:13 [INFO] switch_core.c:1872 switch_core_session_receive_message() Kill DingaLing/1004j [BREAK] 2007-03-21 12:56:13 [DEBUG] mod_dingaling.c:1182 channel_kill_channel() DingaLing/1004j CHANNEL KILL 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2991 sip_i_state() Channel sofia/XX.XX.XX.XX/kieran entering state [ready] 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2996 sip_i_state() Remote SDP: v=0 o=- 7 2 IN IP4 192.168.2.53 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.2.53 t=0 0 m=audio 37468 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2595 negotiate_sdp() Codec Compare [PCMU:0]/[PCMU:0] 2007-03-21 12:56:15 [INFO] mod_sofia.c:1578 tech_set_codec() Set Codec sofia/XX.XX.XX.XX/kieran PCMU/8000 20 ms 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2571 negotiate_sdp() Set 2833 dtmf payload to 101 2007-03-21 12:56:15 [INFO] mod_sofia.c:1635 activate_rtp() RTP [sofia/XX.XX.XX.XX/kieran] XX.XX.XX.XX:16386->192.168.2.53:37468 codec: 0 ms: 20 2007-03-21 12:56:15 [DEBUG] switch_rtp.c:487 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2007-03-21 12:56:15 [NOTICE] mod_sofia.c:3247 sip_i_state() Channel [sofia/XX.XX.XX.XX/kieran] has been answered 2007-03-21 12:56:15 [DEBUG] switch_ivr.c:3074 switch_ivr_originate() Originate Resulted in Success: [sofia/XX.XX.XX.XX/kieran] The weird thing is that if I call the same extension in default_context.xml using a sip phone registered locally it bridges without any problem to the remote sip registration. Can anyone please help with this? I've been battling it for a long time now. I've learned a lot which is good though :-). By the way I just downloaded the last svn version today, so I couldn't be on a more recent version :-) Also if this isn't the best place to address this type of question if anyone could point me in the right direction that would be greatly appreciated. Thanks for any help. Kieran.

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two problems with mod_conference
by Mark D. Anderson 21 Mar '07

21 Mar '07

I'm currently running head (4687) but these problems have been unchanged since I've started playing with mod_conference about 2 months ago. I have tried a cross-product of two ITSPs (vitelity and teliax) either landline caller or cell caller mod_conference "native", or a javascript based on confroom.js I have found some differences in behavior in these 8 cases. 1. Missing 1 second of initial audio (callers don't hear the beginning of the "please enter your pin" message). It doesn't matter whether I move the session.answer() earlier. This happens except for {teliax, landline, *} 2. Doubling (and tripling) of incoming dtmf digits. It is quite hard to get it to accept a pin, because it keeps thinking I've entered a digit twice, when I haven't. As far as I can see, there is no user configuration to tune dtmf configuration (like min/max length of a tone, or echo cancellation). I also don't know if there is a way to force freeswitch/sofia to use only a particular dtmf system. Note that the problem seems worse when running javascript confroom.js, but it happens both ways. This happens with both ITSPs. I see it except for {*, landline, native mod_conference.c}. Any suggestion for causes, fixes, ways to diagnose further, etc., are welcome. -mda

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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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