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Re: [Freeswitch-dev] question about codecs negotiation
by Anthony Minessale 01 Nov '09

01 Nov '09

I meant more so that we approved of any patches to existing code when I said standards. If you follow the style used in the other files it is fine. On Oct 31, 2009 7:41 PM, "Georgiewskiy Yuriy" <bottleman(a)icf.org.ru> wrote: On 2009-10-31 19:08 -0500, Anthony Minessale wrote freeswitch-dev(a)lists.fre. ..: AM>My numbered list was a list of facts not an attempt to argue. It is what it AM>is for now and I explained it. I am glad to hear you are interested in AM>helping with the project. We need more people like yourself. AM> AM>FS itself is not only sip based but specifically proxy media and bypass AM>media currently are only sip as they were invented to solve problems in sip AM>at a time when no other existing module could benifit. It would be great if AM>you can get it to work with others. My only requirement is you do not break AM>anthing that already works. Visit us on irc during weekdays or pose your AM>questions in an email and I will try my best to answer when I have time. oк, i think i will have some drafts around a week later, i show it to you to hear you opinion about this changes. AM>On Oct 31, 2009 6:11 PM, "Georgiewskiy Yuriy" <bottleman(a)icf.org.ru> wrote: AM> AM>On 2009-10-31 16:53 -0500, Anthony Minessale wrote freeswitch-dev(a)lists.fre...: AM> AM>AM>4) if you are wiling to enhance the code yourself and it meets our AM>standards AM>AM>you are welcome to work on it ... AM> AM>during for this, where i can read about our standards? and what about standards? i only find some docs on wiki - http://wiki.freeswitch.org/wiki/Documentation/Developer_Documentation , i already read it while develop mod_h323. C уважением With Best Regards Георгиевский Юрий. Georgiewskiy Yuriy +7 4872 711666 +7 4872 71166... _______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org

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Re: [Freeswitch-dev] question about codecs negotiation
by Anthony Minessale 01 Nov '09

01 Nov '09

My numbered list was a list of facts not an attempt to argue. It is what it is for now and I explained it. I am glad to hear you are interested in helping with the project. We need more people like yourself. FS itself is not only sip based but specifically proxy media and bypass media currently are only sip as they were invented to solve problems in sip at a time when no other existing module could benifit. It would be great if you can get it to work with others. My only requirement is you do not break anthing that already works. Visit us on irc during weekdays or pose your questions in an email and I will try my best to answer when I have time. On Oct 31, 2009 6:11 PM, "Georgiewskiy Yuriy" <bottleman(a)icf.org.ru> wrote: On 2009-10-31 16:53 -0500, Anthony Minessale wrote freeswitch-dev(a)lists.fre...: AM>4) if you are wiling to enhance the code yourself and it meets our standards AM>you are welcome to work on it ... during for this, where i can read about our standards? C уважением With Best Regards Георгиевский Юрий. Georgiewskiy ... _______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org

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question about codecs negotiation
by Georgiewskiy Yuriy 31 Oct '09

31 Oct '09

Hi, i already post it in freeswitch-users but nobody reply, now try there. I have questions about fs and not sip protocols, call scheme like this - user1-sip->fs-sip->user2, as i see if in sip profile enable inbound-proxy-media SDP is goes end-to-end if sip is on both call legs, if call goes throuch another protocol module, for example my mod_h323 (call scheme like user1-sip->fs-h323->someH323Ep) there is no way to negotiate codec because i have no SDP from called party for example and have no way to insert it on sip leg, or i don't see this way? This issue slow down my development, especially t38 part becouse i don't known how to resolve this in true way. C уважением With Best Regards Георгиевский Юрий. Georgiewskiy Yuriy +7 4872 711666 +7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО "Ай Ти Сервис" IT Service Ltd http://nkoort.ru http://nkoort.ru JID: GHhost(a)jabber.tula-ix.net.ru JID: GHhost(a)jabber.tula-ix.net.ru YG129-RIPE YG129-RIPE

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FreeSWITCH Weekly Conference Call Starting Shortly, Please Start Calling In
by Michael Collins 30 Oct '09

30 Oct '09

Hey everyone! The weekly meeting will start in a few minutes. Here's the agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_30 Please feel free to add other subjects. If you have something to share with the community that would be great. Also, if you have questions for the dev team by all means bring them too. -Michael

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celt codec update
by vip killa 30 Oct '09

30 Oct '09

could someone help me understand how to compile freeswitch with the new development release of celt (0.7.0) ?

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building a module
by Tihomir Culjaga 29 Oct '09

29 Oct '09

can someone help on this. I'm trying to build the module on trunk and it looks like i can't find 'switch_strlen_zero' function anywhere :P making all mod_rad_auth Compiling utils.c... Compiling mod_rad_auth.c... quiet_libtool: compile: gcc -I/home/tculjaga/freeswitch-trunk/src/include -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_rad_auth.c -fPIC -DPIC -o .libs/mod_rad_auth.o cc1: warnings being treated as errors mod_rad_auth.c: In function ‘rad_auth_function’: mod_rad_auth.c:301: error: implicit declaration of function ‘switch_strlen_zero’ make[4]: *** [mod_rad_auth.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_rad_auth-all] Error 1 make[1]: *** [mod_rad_auth] Error 2 on an older revision works well ... [tculjaga@l01sipindir2 freeswitch]$ svn info Path: . URL: http://svn.freeswitch.org/svn/freeswitch/trunk Repository Root: http://svn.freeswitch.org/svn Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2 Revision: 14640 Node Kind: directory Schedule: normal Last Changed Author: intralanman Last Changed Rev: 14640 Last Changed Date: 2009-08-26 17:11:07 +0200 (Wed, 26 Aug 2009) [tculjaga@l01sipindir2 freeswitch]$ [tculjaga@l01sipindir2 freeswitch]$ [tculjaga@l01sipindir2 freeswitch]$ [tculjaga@l01sipindir2 freeswitch]$ sudo make mod_rad_auth making all mod_rad_auth Compiling utils.c... Compiling mod_rad_auth.c... mkdir .libs Compiling mod_rad_auth.c ... Creating mod_rad_auth.so... Can someone please advice? T.

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Network cable disconnection
by Suneel Papineni 29 Oct '09

29 Oct '09

Hi, I have configured FreeSwitch and played it by making calls. Every thing went fine. Then I tried to see what happens if network cable is unplugged. Freeswitch shown an error saying ping failed to the gateway and IP address is changed to local system IP. After sometime network cable plugged back in and in another 30 seconds freeswitch has shown it got the IP of gateway and changed back from local IP to gateway IP. After that when I tried to make calls, calls are not going through. Even there is no log displayed in Freeswitch. Please find below the logs... freeswitch@QA360BASExp> 2009-10-28 16:46:13.156250 [WARNING] sofia.c:2814 Ping failed ihub-trunk 2009-10-28 16:46:13.234375 [INFO] mod_sofia.c:3160 IP change detected [192.168.12.10]->[192.168.20.138] []->[] 2009-10-28 16:46:13.265625 [NOTICE] sofia_glue.c:3444 Reload XML [Success] 2009-10-28 16:46:13.265625 [INFO] switch_time.c:661 Timezone reloaded 0 definitions 2009-10-28 16:46:14.156250 [NOTICE] sofia.c:921 Waiting for worker thread 2009-10-28 16:46:14.156250 [NOTICE] sofia_glue.c:3505 deleted gateway ihub-trunk 2009-10-28 16:46:14.156250 [NOTICE] sofia_reg.c:2122 Added gateway 'ihub-trunk' to profile 'TrunkExternal' 2009-10-28 16:46:14.156250 [NOTICE] sofia.c:2767 Started Profile TrunkExternal [sofia_reg_TrunkExternal] 2009-10-28 16:46:14.156250 [ERR] sofia.c:805 Error Creating SIP UA for profile: TrunkExternal 2009-10-28 16:46:14.156250 [NOTICE] sofia_glue.c:3505 deleted gateway ihub-trunk 2009-10-28 16:46:53.234375 [INFO] mod_sofia.c:3160 IP change detected [192.168.20.138]->[192.168.12.10] []->[] 2009-10-28 16:46:53.281250 [NOTICE] sofia_glue.c:3444 Reload XML [Success] 2009-10-2816:46:53.281250 [INFO] switch_time.c:661 Timezone reloaded 0 definitions --After this there were no messages Calls went through if I reload freeswitch. Another surprise thing is I checked for the active ports in the system before and after cable unplugged. Port 5060 disappeared after cable is plugged out. This doesn't display even after cable is plugged back in. 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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Registering a sip phone with 'Multi companies' FS server
by Peter Volchek 28 Oct '09

28 Oct '09

Hi, I am trying to build a Multi Companies' environment, as described at 'http://wiki.freeswitch.org/wiki/Multiple_Companies' On my desk I have a Polycom sip phone, which I was registering with my fs server as 1004. (using a default profile) Now, I want my Soft-Phone, to be a number but belonging to another company (I made a profile test.com) say 5059 However, when I ty to register it with FS, I am getting freeswitch@internal> 2009-10-28 11:57:57.882850 [WARNING] sofia_reg.c:1731 Can't find user [5059(a)fs.xxx.com] You must define a domain called 'fs.xxx.com' in your directory and add a user with the id="5059" attribute and you must configure your device to use the proper domain in it's authentication credentials. I use X-Lite soft phone (counterpath), and I do not see any settings to give FS a clue, to look inside test.com domain. I am specifying: Display Name: My soft phone User Name: 5059 Password: 5059 Authorization name: 5059 Domain: fs.xxx.com (FS is installed there) [v] Register with domain and receive incoming calls (*) Send outbound via: domain In other words, is it possible to register the phones (from different companies) with FS with Multi-Company configuration? What did I miss? Cheers, PV

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Using a modem to connect ordinary phones
by Alex To 28 Oct '09

28 Oct '09

Hi, I found the same topic created 2 years back in Oct 2007 but it doesn't seem to have the answer yet. I am quite new to FS and having a hard time reading the documentation. Someone could tell me: how would I use a modem and develop an Endpoint in FS that I could bridge the call from VoIP to PSTN. I am a hardware engineer, I am developing this modem and would want to use it with FS. I know that I could simply buy a VoIP Gateway or a zaptel compatible card and use OpenZap. Could someone point me out the direction for the following matters: 1. I feel that I need to develop an Endpoint module, like mod_sofia, mod_iax... that controls my modem so I could bridge the call from VoIP to PSTN ? If that is correct, could someone point me the documentation to start reading about Endpoint interface .? 2. Is there any similar module developed before about controlling a modem using AT Command Pardon my bad English, I don't come from an English speaking country. Btw, this is the first time I participate in a mailing list so I still have no idea how to create a thread or reply to a posting like I normally do in forums. Thanks very much and any helps are highly appreciated. Regards Alex To

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error in loading spidermonkey
by Erwin Davis 28 Oct '09

28 Oct '09

Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04 runs on fedora 8 VM. Any clue? thanks, [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**

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