- FreeSWITCH-users - FreeSWITCH Mailing Lists

Sending Out Event from JS
by Pete Kay 27 Apr '08

27 Apr '08

Hi, I am trying to get events to be fired out using the following codes in JS: e = new Event("custom", "message"); var txt = "main menu==========="; e.addBody(txt); session.sendEvent(e); Within my client, I sent "Plain event all" to listen to all events. Then, I filtered on all the events, but I can't find anything that has the text "main", "menu", "custom" or "message". I am wondering if my code has bug or I was not looking at the right place in my event log. Could someone please give me some suggestion? Any help will be greatly appreciated. Thanks, Pete

1 0

Call For Open Source Awards 2008 Nominations
by Brian West 26 Apr '08

26 Apr '08

http://radar.oreilly.com/archives/2008/04/open-source-award-nominations.html you know what to do! /b

2 1

eavesdrop dtmf commands
by Tamas Cseke 26 Apr '08

26 Apr '08

Hello, Could you please tell me the documented behaviour is correct on this page: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop When I eavesdrop an uuid without any dtmf command: The other half is deaf, and I can talk with uuid, and we are hear the other half. 0 and 1 dtmfs seems to be inefficient, nothing change, the behaviour is the same as above. But when i try with dtmf 2, 3 or three_way app. the uuid hear a terrible noise. I tried this with svn trunk r8193 both on windows xp and ubuntu linux. I would like to make somehow a three way call. And somehow uuid should talk with the other half during spy, could you tell me if is it possible? Thanks any help, Tamas

2 2

Having problem sending and receiving faxes
by Pete Kay 26 Apr '08

26 Apr '08

Hi, I am unable to get FS to send and receive faxes and I hope someone can give me a hand. When sending the fax, I get the following error from sock2me: Sending Fax filename: [/home/anne/abcd.tiff] from 127.0.0.1:9015 -> 127.0.0.1:9016 Phase B handler on session 0 - (0x80) DIS Phase E handler on session 0 - (14) TIFF/F file cannot be opened Phase E: bit rate 9600 Phase E: ECM on Phase E: pages transferred 0 Phase E: image size 0 x 0 Phase E: image resolution 0 x 0 Phase E: bad rows 0 Phase E: longest bad row run 0 Phase E: coding method ??? Phase E: image size 0 bytes Phase E: local ident 'Socket 2 ME' Phase E: remote ident '' Phase E: Remote was made in 'Japan' Phase E: Remote was made by 'Sharp/Olivetti' Done There should not be problem with permission, as I can do the following: ser:/usr/local/freeswitch# ls -al /home/anne/abc.tiff -rw-r--r-- 1 anne anne 15344 2008-04-26 22:06 /home/anne/abc.tiff When attempting to send from FS, there is no response from sock2me. In the console, I get the following: 2008-04-27 08:41:30 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1006->fax@all 2008-04-27 08:41:30 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test conditions destination_number(fax) =~ /^fax$/ 2008-04-27 08:41:30 [DEBUG] switch_core_state_machine.c:96 switch_core_standard_on_ring() sofia/default/1006@192.168.1.104:5060 State Change CS_RING -> CS_EXECUTE 2008-04-27 08:41:30 [DEBUG] switch_core_session.c:670 switch_core_session_signal_state_change() Kill sofia/default/ 1006@192.168.1.104:5060 [BREAK] 2008-04-27 08:41:30 [DEBUG] switch_core_state_machine.c:460 switch_core_session_run() (sofia/default/1006@192.168.1.104:5060) State RING going to sleep 2008-04-27 08:41:30 [DEBUG] switch_core_state_machine.c:406 switch_core_session_run() sofia/default/1006@192.168.1.104:5060 Running State Change CS_EXECUTE 2008-04-27 08:41:30 [DEBUG] switch_core_state_machine.c:470 switch_core_session_run() (sofia/default/1006@192.168.1.104:5060) State EXECUTE 2008-04-27 08:41:30 [DEBUG] mod_sofia.c:156 sofia_on_execute() sofia/default/1006@192.168.1.104:5060 SOFIA EXECUTE 2008-04-27 08:41:30 [DEBUG] switch_core_state_machine.c:129 switch_core_standard_on_execute() Standard EXECUTE 2008-04-27 08:41:30 [DEBUG] switch_core_state_machine.c:142 switch_core_standard_on_execute() sofia/default/1006@192.168.1.104:5060Execute answer() 2008-04-27 08:41:30 [DEBUG] switch_core_state_machine.c:142 switch_core_standard_on_execute() sofia/default/1006@192.168.1.104:5060Execute echo(get a fax=======================>) 2008-04-27 08:41:30 [DEBUG] switch_core_io.c:347 switch_core_session_read_frame() Engaging Read Buffer at 480 bytes 2008-04-27 08:41:58 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_outbound] status [102][NAT binding changed] session: n/a 2008-04-27 08:41:58 [DEBUG] sofia.c:345 sofia_event_callback() nua_i_outbound: unknown event 8: 102 NAT binding changed 2008-04-27 08:41:58 [DEBUG] sofia_reg.c:87 sofia_reg_check_gateway() registered callwithus 2008-04-27 08:42:01 [DEBUG] sofia.c:1565 sofia_handle_sip_i_state() Channel sofia/default/1006@192.168.1.104:5060 entering state [terminated] 2008-04-27 08:42:01 [NOTICE] sofia.c:1962 sofia_handle_sip_i_state() Hangup sofia/default/1006@192.168.1.104:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2008-04-27 08:42:01 [DEBUG] switch_channel.c:1355 switch_channel_perform_hangup() Kill sofia/default/1006@192.168.1.104:5060[KILL] 2008-04-27 08:42:01 [DEBUG] switch_core_session.c:670 switch_core_session_signal_state_change() Kill sofia/default/ 1006@192.168.1.104:5060 [BREAK] 2008-04-27 08:42:01 [DEBUG] switch_core_state_machine.c:470 switch_core_session_run() (sofia/default/1006@192.168.1.104:5060) State EXECUTE going to sleep 2008-04-27 08:42:01 [DEBUG] switch_core_state_machine.c:406 switch_core_session_run() sofia/default/1006@192.168.1.104:5060 Running State Change CS_HANGUP 2008-04-27 08:42:01 [DEBUG] switch_core_state_machine.c:434 switch_core_session_run() (sofia/default/1006@192.168.1.104:5060) State HANGUP 2008-04-27 08:42:01 [DEBUG] mod_sofia.c:255 sofia_on_hangup() Channel sofia/default/1006@192.168.1.104:5060 hanging up, cause: NORMAL_CLEARING 2008-04-27 08:42:01 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() Standard HANGUP sofia/default/ 1006@192.168.1.104:5060, cause: NORMAL_CLEARING 2008-04-27 08:42:01 [DEBUG] switch_core_state_machine.c:434 switch_core_session_run() (sofia/default/1006@192.168.1.104:5060) State HANGUP going to sleep 2008-04-27 08:42:01 [DEBUG] switch_core_session.c:730 switch_core_session_thread() Session 160 (sofia/default/ 1006@192.168.1.104:5060) Locked, Waiting on external entities 2008-04-27 08:42:01 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 160 (sofia/default/ 1006@192.168.1.104:5060) Ended 2008-04-27 08:42:01 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/ 1006@192.168.1.104:5060 [CS_HANGUP] 2008-04-27 08:42:01 [DEBUG] switch_core_media_bug.c:284 switch_core_media_bug_remove_all() Removing BUG from sofia/default/ 1006@192.168.1.104:5060 Can someone please kindly tell me what the problem could be and how can I fix it? Thanks alot for all you kind input. Best Regards, Pete

1 0

Problems with initial setup - basic nat
by Jay Reeder 26 Apr '08

26 Apr '08

We're setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. I've wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal. I've been through the wiki and reviewed list archives but I'm missing something. I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples. This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewall's external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch. This is where I'm stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch won't accept the call and is logging: 2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled. I'm sure it's something obvious but what am I missing? Thanks, Jay

5 12

make install error
by Angelito Manansala 26 Apr '08

26 Apr '08

Hi Guys, I've been playing with FS a week ago, I'm trying to install wikipbx following their installation instruction wiki now I have a error on make install command, below are some portions of the logs. mod_cepstral.c:379: error: 'cepstral_t' has no member named 'port' mod_cepstral.c: In function 'cepstral_numeric_param_tts': mod_cepstral.c:389: error: 'cepstral_t' has no member named 'port' mod_cepstral.c: In function 'cepstral_float_param_tts': mod_cepstral.c:401: error: 'cepstral_t' has no member named 'port' mod_cepstral.c: In function 'mod_cepstral_load': mod_cepstral.c:410: error: 'engine' undeclared (first use in this function) make[5]: *** [mod_cepstral.o] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_cepstral-install] Error 1 make[2]: *** [install-recursive] Error 1 make[1]: *** [install-recursive] Error 1 make: *** [install] Error 2 -- Regards, Lito

2 1

A small question about Freeswitch security
by Krzysiek 26 Apr '08

26 Apr '08

Hi once again :) I have a small question, because I don't know whether I understand Freeswitch security things correctly... If uncomment this line in the configs: <param name="Inbound-no-media" value="true"> and enable SRTP encryption (key via SDES or MIKEY) in phones, will I have to have support for those SDES or MIKEY in Freeswitch ? I thnik that sdp messages within sip will be passed through Freeswitch unchanged and i don't have to have support for those in Freeswitch and the call will be secure. I'm writing about private network, where calls are routed only between endpoionts in this private network. It is a theoretical question and I know that Freeswitch doesn't have MIKEY support. sorry for my english, I hope you understand me. Chris

2 1

Hung Phone
by Luis Jimenez 26 Apr '08

26 Apr '08

Hi, i have this in default.xml <extension name="*99"> <condition field="destination_number" expression="^\*99$" break="always"> <action application="set" data="myvar=1004"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/default/${myvar}%10.0.0.100"/> </condition> </extension> when i call, it works, but when i hangup the extension 1004 (callee), my caller phone stays connected until i hagup manually this one. if i hangup the caller first the calee hangup fine. Luis jimenez

3 5

Syntax for SIP2Jingle Call in Dialplan
by Chris Chen 25 Apr '08

25 Apr '08

I have setup one jabber server running along the Freeswitch on the centos 5, and setup the gtalk in public context as the jabber server component on port 5347 , and established the federation with googletalk network. Now I can call from gtalk client to my sip account on freeswitch without problem by the following dialplan: <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^conf\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/default/$1%192.168.1.253"/> </condition> </extension> where 192.168.1.253 is the ip address of freeswitch. but I am having issues with the sip2jingle call. in the default context dialplan, I can do this <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^1500$"> <action application="bridge" data="dingaling/ jabber.mydomain.com/mybuddy@gmail.com"/> </condition> </extension> where the jabber.mydomain.com is the component profile name I defined for the gtalk server component. but if I modify the dialplan to do more general sip2jingle call as below: <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/ jabber.mydomain.com/$1@gmail.com"/> </condition> </extension> or <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@ ${sip_fromhost}/$1(a)gmail.com"/> </condition> </extension> I tested with jingle+mybuddy.lastname(a)gmail.com<jingle%2Bmybuddy.lastname(a)gmail.com>from my extension 1000 on soft phone Bria 2.2 on XP, I got the error " 1000(a)gmail.com not found" Could you guys share the correct syntax for doing SIP to Jingle calls? Thanks, Chris

1 0

Shall we revive the ailing mod_cdr?
by Yossi Neiman 25 Apr '08

25 Apr '08

Hi folks, It's been quite a while. Some of you might know of the mod_cdr that was fully functional up until around sometime after July 2007. Due to personal issues, I did not have time to maintain it. I am now in a position to revive it. I will be making some changes to how it is designed, and removing some of the serialization. Currently in FreeSWITCH we have the following CDR logging formats at hand (separate of mod_cdr): Individual XML flat files, comma separate values, and curl HTTP handling. As such there is no need to include these in mod_cdr anymore, but I am looking for what backends everybody would be interested in. I will be starting off with the ODBC backend. Please note that database-specific backends are possible, but there is a possibility they would not be available for inclusion in trunk. Another format that I thought just for the fun of it would maybe to shoot out CDRs as e-mails (donno if that's at all useful...). Give me your thoughts and ideas. -- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com

3 4