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please, help: invalid bind URL
by Oleg Khovayko 25 Jun '13

25 Jun '13

Hi, I am using relative old FreeSWITCH (~2 years old), it worked good for a long time. Today, suddenly, it stop work, whit message: /2013-06-18 10:59:33.163240 [ERR] sofia.c:2199 Error Creating SIP UA for profile: internal// //2013-06-18 10:59:33.164289 [ERR] sofia.c:2199 Error Creating SIP UA for profile: external// / I collected sofia debug info, got: sofia loglevel all 9 Sofia log level for component [all] has been set to [9] freeswitch(a)olegh.ath.cx> sofia profile internal start Reload XML [Success] internal started successfully 2013-06-18 11:39:37.314934 [NOTICE] sofia.c:5057 Started Profile internal [sofia_reg_internal] 2013-06-18 11:39:37.375810 [INFO] mod_enum.c:871 ENUM Reloaded 2013-06-18 11:39:37.375810 [INFO] switch_time.c:1163 Timezone reloaded 530 definitions nua: nua_create: entering nua: nua_stack_init: entering nua: nua_stack_set_params: entering soa_create("default", 0x289a1dc0, 0x28c8f6a0) called soa_set_params(static::0x28903350, ...) called soa_set_params(static::0x28903350, ...) called nta_agent_create: initialized hash tables nta_agent_create: initialized transports nta_agent_create: initialized random identifiers nta_agent_create: initialized timer nta_agent_create: initialized resolver *nta: sip:mod_sofia@:5060;maddr=192.168.1.5: invalid bind URL* nua: initializing SIP stack failed nua: nua_stack_deinit: entering sres_sofia_update(0x289a0220, -1, -1) sres_sofia_update(0x0, -1, -1) tport_destroy(0x0) 2013-06-18 11:39:37.485116 [ERR] sofia.c:2199 Error Creating SIP UA for profile: internal Program run parameters: /usr/local/freeswitch/bin/freeswitch -u sip -g sip -waste -hp -nonat OS: FreeBSD v 7.2 This machine behind Verizon Fios router, NATed, static IP=192.168.1.1 Please help, if possible. Oleg

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FreeSWITCH for WebRTC
by Richard Screene 24 Jun '13

24 Jun '13

Is there any news on when WebRTC support will be added to FreeSWITCH? It would be awesome if we could host a video conference using FreeSWITCH. Regards, Richard

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Odp: Re: [Freeswitch-users] Github and FreeSWITCH
by Mariusz Czulada 21 Jun '13

21 Jun '13

On our wiki there is a section: http://wiki.freeswitch.org/wiki/Git_Tips#Github_Workflow I take this information as a sort of incentive to use this alternative way to publish patches. Perhaps this section should be removed? Regards, Mariusz Dnia Piątek, 21 Czerwca 2013 15:22 Brian West <brian(a)freeswitch.org> napisał(a) > I will investigate that... just wanted to make sure everyone was aware we use Jira and our wiki so we don't have fragmentation. > > On Jun 21, 2013, at 8:13 AM, Mathieu Parent <math.parent(a)gmail.com> wrote: > > > Hi, > > > > 2013/6/21 Brian West <brian(a)freeswitch.org>: > >> Please use http://jira.freeswitch.org to publish patches. > > > > Then, why don't you simply remove the ability to request pull/fill > > issues/update wiki? > > > > Regards > > -- > > Mathieu Parent > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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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store message thread for handling extra register.
by ThanhTruong 20 Jun '13

20 Jun '13

Hi all, I have small issue when i am trying to handle extra register with another server. Let me explain my idea. First : when xlite(192.168.1.158) send REGISTER message to FS, it will store message (or something like that to recognize thread from xlite, but I dont know exactly which value will be stored in msg1). Then, I will send one REGISTER message to another server (like asterisk (192.168.1.174)), and when asterisk server return(100 first, then 401 include realm, nonce, auth_params), I need to store message thread like the first one (to reused like msg2), then, i will use nua_response to transfer 401 message to xlite(with realm, nonce). Then I can do the same when i get REGISTER message (with password in md5) from xlite and forwarding to asterisk server for authen. So, I have try to store message in to msg_t variable but i get error 500 - Responding to a Non-Existing Request when try to send back 401 with realm, nonce to xlite. I have used this function to store msg to one variable: (in function sofia.c/sofia_event_callback ) if(event == nua_i_register){ msg1 = de->data->e_msg ; switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "tnt: done store msg1 into catche \n"); } if (event == nua_r_register){ msg2 = de->data->e_msg ; switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "tnt: done store msg2 into catche \n"); } and send back 401 like: case nua_r_register: switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "nua_r_register : process handler ===========>>>>> \n "); if (status == 401 || status == 407){ sip_www_authenticate_t const* www_auth = sip->sip_www_authenticate; char const* scheme = www_auth->au_scheme; const char* realm = msg_params_find( www_auth->au_params, "realm=" ); const char* nonce = msg_params_find( www_auth->au_params, "nonce=" ); char *auth_str; auth_str = switch_mprintf("%s realm=\"%q\", nonce=\"%q\",%s algorithm=MD5, qop=\"auth\"", scheme,realm, nonce," stale=true,"); nua_respond(nh, SIP_401_UNAUTHORIZED, NUTAG_WITH_THIS_MSG(msg2), SIPTAG_WWW_AUTHENTICATE_STR(auth_str), TAG_END()); break; } break; I am new in this and need your help, please suggest or reference some link that i can get it work, thank all in advance. ThanhTruong

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FreeSWITCH Community Conference Call - ClueCon 2013 Schedule!
by Michael Collins 19 Jun '13

19 Jun '13

Hello all! This is a reminder that we will be releasing the ClueCon 2013<http://www.cluecon.com>schedule on today's community conference call <http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_19>. We also have the whole FreeSWITCH core development team joining us today. The call starts at 1PM Eastern, 10AM Pacific. We look forward to having a large crowd and some stimulating conversation on today's call! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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CHANGED Skype gw to Weekly Conference Call
by Giovanni Maruzzelli 19 Jun '13

19 Jun '13

Hey FreeSWITCHers, while I was testing performance related innovation I invalidated by mistake the Skype account you called to connect to the Weekly Call. So, the "skypiax5" account you used to call is no more. Please call skype "freeswitch_conf_call" to be automatically connected to the Weekly Call. Sorry for the inconvenience, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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FreeSWITCH Weekly News and Notes
by Michael Collins 17 Jun '13

17 Jun '13

Greetings! We are all looking forward to our upcoming conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_19>for June 19th, because we are releasing the official ClueCon 2013 <http://www.cluecon.com> speaker schedule! Additionally, the FreeSWITCH core developer team will be on the conference call to hang out with us. Let's all be there this Wednesday and pack out the conference call! In addition to FreeSWITCH, the core development team also built the Cudatel Communication Server <http://www.cudatel.com>. On last week's conference call <http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_12> we did a thorough tour of the CudaTel interface and its capabilities. We also talked about how the CudaTel can utilize external call control by utilizing an HTTAPI server. The CudaTel really is a great product and we highly recommend it for those enterprises that need a simple yet powerful IP communications platform. Thanks for your time and we'll talk to everyone on Wednesday! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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mod_fifo
by Thierry Panthier 17 Jun '13

17 Jun '13

Hello everyone, When a call leaves mod_fifo (and it gets bridged) if I do a *fifo list <myqueue>*, I get: <bridges> <bridge fifo_name="emergency" bridge_start="2013-06-12 07:45:03" bridge_start_epoch="1371023103"> <caller uuid="a16767af-e252-43b5-ad33-cbd38a42bc64" caller_id_name=* "Outbound%20Call"* caller_id_number=*"XML"*></caller> <consumer> <uuid>4b84ab5e-f2a2-4d87-8295-7726663abeed</uuid> <outgoing_uuid>0b9fa586387efe4014e010bada4da9cc</outgoing_uuid> </consumer> </bridge> </bridges> However I need to get the *caller_id_name* and the *caller_id_number* of the entity that initiated the call and not "*Outbound Call*" or *"XML"*. I read in a thread that I should set *origination_callee_id_name* and * origination_callee_num* but how do I do that? I tried to set a variable in my dialplan and use it in *fifo.conf.xml *but I get the following error: *[CRIT] switch_channel.c:1364 Invalid data (${origination_callee_id_name} contains a variable)* Here's my *original* fifo.conf.xml: <fifos> <fifo name="emergency" importance="0"> <member timeout="60" simo="1" lag="0">{member_wait=nowait}loopback/dequeue_emergency/default/XML</member> </fifo> </fifos> Here's part of my *original* dialplan: * * <extension name="queue_emergency"> <action application="set" data="fifo_music=$${hold_ringing}"/> <action application="set" data="fifo_orbit_exten=timeout_emergency:$${emergency_queue_timeout}"/> <action application="answer"/> <action application="fifo" data="emergency in"/> </extension> Here's my *modified* fifo.conf.xml: <fifos> <fifo name="emergency" importance="0"> <member timeout="60" simo="1" lag="0">{member_wait=nowait,origination_callee_id_name=${some_var_set in_the_dialplan}}loopback/dequeue_emergency/default/XML</member> </fifo> </fifos> If I set to a fixed value instead of a variable, it works fine. But that's useless to me. Any help would be greatly appreciated. Regards, Thierry Panthier

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New Wiki Page: "Reloading" (re-transmission)
by Michael Collins 16 Jun '13

16 Jun '13

FYI, Per Steven's mention on the other thread about not knowing when to reloadxml vs. restart freeswitch vs. <whatever command> I promised to add a new wiki page. Here it is: http://wiki.freeswitch.org/wiki/Reloading I've seeded the page with some explanatory text and a few items in the table. Please add to it with what you've come across in your usage of FreeSWITCH. Thanks, Michael -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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New Wiki Page: "Reloading"
by Michael Collins 16 Jun '13

16 Jun '13

FYI, Per Steven's mention on the other thread about not knowing when to reloadxml vs. restart freeswitch vs. <whatever command> I promised to add a new wiki page. Here it is: -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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