May 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

This weeks Meeting
by Brian West 13 May '08

13 May '08

http://wiki.freeswitch.org/wiki/Wiki_meet_2008_05_14 Please add to the agenda ASAP. We'll be going over everything in Jira this week for sure.... http://jira.freeswitch.org Hope to hear you on the call! ;) /b

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problems with outgoing calls
by Czaderna 13 May '08

13 May '08

Hello :) At the begining - sorry for my english. I have following problems with freeswitch: 1. I have got 3 computers in LAN. On one computer is running freeswitch, and on the others are intalleted Xlite software (sip clients). 2. On "voipdiscount.com" I've got an account for free local calls. 3. I want make a call on mobile phone using freeswitch and voipdiscount.com So on my computer I start XLite, log in on default freeswitch account 1000 and then try dial a mobile phone number 4. On dialplan I have something like this: ./CONF//DIALPLAN: public.xml <include> <context name="public"> <extension name="public_extensions"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> <extension name="test_voipdiscount"> <condition field="destination_number" expression="^([0-9]{13})$"> <action application="bridge" data="sofia/gateway/voipdiscount.com/$1@sip.voipdiscount.com"/> </condition> </extension> </context> </include> and on the sip_profiles (conf/sip_profiles/outbound/voipdiscount.com.xml) I have following configuration of gateway: <include> <gateway name="voipdiscount.com"> <param name="username" value="user_name"/> <param name="realm" value="sip.voipdiscount.com"/> <param name="password" value="user_password"/>--> <param name="proxy" value="sip.voipdiscount.com"/> </gateway> </include> After connection on freeswitch I get following messages: 2008-05-13 09:51:43 [INFO] mod_iax.c:928 mod_iax_runtime() IAX Ready Port 4569 freeswitch@xx> 2008-05-13 09:51:43 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering voipdiscount.com 2008-05-13 09:51:43 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering ideasip.com 2008-05-13 09:51:47 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1000(a)192.168.3.79 [701574c2-20c1-11dd-9725-39613e150eca] 2008-05-13 09:51:47 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1000->0048606xxxxxx@public 2008-05-13 09:51:47 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/outbound/0048606xxxxxx(a)sip.voipdiscount.com [70165892-20c1-11dd-9725-39613e150eca] 2008-05-13 09:52:14 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering voipdiscount.com 2008-05-13 09:52:14 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering ideasip.com 2008-05-13 09:52:46 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering voipdiscount.com 2008-05-13 09:52:46 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering ideasip.com 2008-05-13 09:52:47 [NOTICE] switch_ivr_originate.c:1154 switch_ivr_originate() Hangup sofia/outbound/NTK(a)sip.voipdiscount.com [CS_HOLD] [NO_ANSWER] 2008-05-13 09:52:47 [INFO] mod_dptools.c:1551 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL 2008-05-13 09:52:47 [NOTICE] mod_dptools.c:1578 audio_bridge_function() Hangup sofia/default/1000(a)192.168.3.79 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2008-05-13 09:52:47 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 1 (sofia/default/1000(a)192.168.3.79 ) Ended 2008-05-13 09:52:47 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/1000(a)192.168.3.79 [CS_HANGUP] 2008-05-13 09:52:47 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 2 (sofia/outbound/0048606xxxxxx(a)sip.voipdiscount.com) Ended 2008-05-13 09:52:47 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/outbound/0048606xxxxxx(a)sip.voipdiscount.com [CS_HANGUP] There are no errors but after few minutes connection is hang up (no calling signal on the channnel). I don't know where is the problem. Also I was trying connect through ideasip.com (got SIP number and an account at this provider) same as above, but in the dialplan instead of "bridge" I used "redirect". So my dialplan looks like this: <include> <context name="public"> <extension name="public_extensions"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> <extension name="test_voipdiscount"> <condition field="destination_number" expression="^([0-9]{11})$">  <action application="redirect" data="sip:$1@proxy.ideasip.com"/>   </condition> </extension> </context> </include> On voipdiscount.com this trick doesn't working. Could anyone help me, please ? Thanks Paul.

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registration failed . can't find user
by sye 12 May '08

12 May '08

Hi, I just installed freeswitch. I use softphone to register user 1005 but registration failed. ( For any user under conf/directory/default directory). This is my directory/default.xml is <include>  <domain name="$${domain}"> <params> <param name="dial-string" value="{transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_domain}/${dialed_ext}@${dialed_domain})}"/> </params> <variables> <variable name="record_stereo" value="true"/> </variables> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> </domain> <domain name="$${sip_profile}"> <user id="stanley"> <params> <param name="password" value="stanley" /> </params> </user> </domain> <domain name="10.3.2.34"> <user id="sye"> <params> <param name="password" value="sye" /> </params> </user> </domain> </include> 2008-05-11 23:49:23 [WARNING] sofia_reg.c:892 sofia_reg_parse_auth() can't find user [1005(a)10.3.2.34] 2008-05-11 23:49:23 [DEBUG] sofia_reg.c:482 sofia_reg_handle_register() send challange for [1005(a)10.3.2.34] 2008-05-11 23:50:16 [WARNING] sofia_reg.c:892 sofia_reg_parse_auth() can't find user [stanley(a)10.3.2.34] 2008-05-11 23:50:16 [DEBUG] sofia_reg.c:482 sofia_reg_handle_register() send challange for [stanley(a)10.3.2.34] 2008-05-11 23:50:55 [DEBUG] sofia_reg.c:482 sofia_reg_handle_register() send challange for [sye(a)10.3.2.34] Any suggestion? Thanks, Stanley

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proxy authentication required
by sye 12 May '08

12 May '08

I install a freeswitch at this ip address 208.74.66.11. I make as ip call from gizmo (using 17473133317(a)proxy01.sipphone.com) to 1000(a)208.74.66.11. I have two question: 1. why freeswitch tried to find user 17473133317(a)208.74.66.11 ? The phone number belongs to siphone.com, not my freeswitch domain, 2. why it sends challange for 1000(a)209.74.66.11 ? 2008-05-01 19:13:55 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a 2008-05-01 19:13:56 [WARNING] sofia_reg.c:892 sofia_reg_parse_auth() can't find user [17473133317(a)208.74.66.11] 2008-05-01 19:13:56 [DEBUG] sofia_reg.c:482 sofia_reg_handle_register() send challange for [1000(a)208.74.66.11] 2008-05-01 19:13:56 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [403][Forbidden] session: n/a I set <param name="accept-blind-reg" value="false"/> in sofia.conf.xml and sip_profile/default.xml and sip_profile/nat.xml. It doesn't help. Can anyone help ? Thanks, Stanley

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Freeswitch using iLBC codec with inphonex
by Leonardo Alves 12 May '08

12 May '08

Hey, I am trying to configure FS to use iLBC with inphonex. But when I try to make a outbound call I get this message: [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] I am changing the codec info in the vars.xml file. <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=iLBC@30i"/> And this global variable is also in the /sip_profile/outbound.xml <param name="codec-prefs" value="$${outbound_codec_prefs}"/> Does anyone know how can I solve this ? Thanks Leonardo Alves

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484
by Jonas Gauffin 12 May '08

12 May '08

Hello Is it possible to get support for response code 484 in mod_xml_curl dialplan? That is to return status code "address incomplete" instead of "not found" in the xml? //Jonas

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Re: [Freeswitch-users] waitForAnswer returns too soon
by Marc Orenberg 10 May '08

10 May '08

Thanks for the help everyone. You were right Anthony. I put the relevant section of the trace below. I also got ahold of a VoicePulse line and tried it, and it worked fine. I really appreciate it! Trace from my Gizmo line: 2008-05-09 11:30:32 [DEBUG] sofia.c:222 sofia_event_callback() event [nua_i_state] status [0][INVITE sent] session: sofia/outbound/17818640108 2008-05-09 11:30:32 [DEBUG] sofia.c:1714 sofia_handle_sip_i_state() Channel sofia/outbound/17818640108 entering state [calling] 2008-05-09 11:30:34 [DEBUG] sofia.c:1714 sofia_handle_sip_i_state() Channel sofia/outbound/17818640108 entering state [ready] 2008-05-09 11:30:34 [DEBUG] sofia.c:1718 sofia_handle_sip_i_state() Remote SDP: v=0 o=root 3158 3158 IN IP4 130.94.88.90 s=session c=IN IP4 130.94.88.90 t=0 0 m=audio 17186 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 2008-05-09 11:30:34 [DEBUG] sofia_glue.c:2024 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2008-05-09 11:30:34 [DEBUG] sofia_glue.c:1333 sofia_glue_tech_set_codec() Set Codec sofia/outbound/17818640108 PCMU/8000 20 ms 160 samples 2008-05-09 11:30:34 [DEBUG] sofia_glue.c:1988 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2008-05-09 11:30:34 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() AUDIO RTP [sofia/outbound/17818640108] 69.64.92.160:26720->130.94.88.90:17186 codec: 0 ms: 20 2008-05-09 11:30:34 [DEBUG] switch_rtp.c:716 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2008-05-09 11:30:34 [NOTICE] sofia.c:2064 sofia_handle_sip_i_state() Channel [sofia/outbound/17818640108] has been answered 2008-05-09 11:30:34 [DEBUG] switch_ivr_originate.c:1213 switch_ivr_originate() Originate Resulted in Success: [sofia/outbound/17818640108] ----- Original Message ---- From: Anthony Minessale <anthony.minessale(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, May 9, 2008 8:49:58 AM Subject: Re: [Freeswitch-users] waitForAnswer returns too soon This block : new_session = new Session(); new_session.originate(session, "sofia/gateway/gizmo/1xxxxxxxxxx"); console_log("info","Waiting for answer!\n"); new_session.waitForAnswer(10000); console_log("info","***ANSWERED\n"); Can be refactored to: new_session = new Session(); console_log("info","Waiting for answer!\n"); new_session.originate(session, "{ignore_early_media=true}sofia/gateway/gizmo/1xxxxxxxxxx"); console_log("info","***ANSWERED\n"); Can be refactored to: console_log("info","Waiting for answer!\n"); new_session = new Session("{ignore_early_media=true}sofia/gateway/gizmo/1xxxxxxxxxx"); console_log("info","***ANSWERED\n"); You may want to compare the console log with debug level (press f8 on cli) and start FS with TPORT_LOG=1 env var set. Many voip providers mistakenly answer calls on the inbound SIP leg before bridging it to the PSTN. You should be able to see the SIP dialog and other console lines telling you if the call was actually answered or not. On Fri, May 9, 2008 at 12:56 AM, Jonathan Palley <jpalley(a)idapted.com> wrote: I think you want: ignore_early_media=true JP On May 9, 2008, at 1:29 PM, Jonas Gauffin wrote: > Doesn't that just allow media to be sent before a 200 ok? In that > case, it doesn't solve the waitForAnswer problem? > > On Fri, May 9, 2008 at 7:05 AM, Leonardo Alves > <leoalves(a)teltec.com.br> wrote: >> Try using the parameter early_media=true >> From: Marc Orenberg >> Sent: Thursday, May 08, 2008 9:39 PM >> To: freeswitch-users(a)lists.freeswitch.org >> Subject: [Freeswitch-users] waitForAnswer returns too soon >> Hello everyone. I'm a new FreeSwitch user, and I'm having a >> problem with >> waitForAnswer() with SpiderMonkey. I'm using it to make an >> outbound call as >> follows: >> >> new_session = new Session(); >> new_session.originate(session, "sofia/gateway/gizmo/ >> 1xxxxxxxxxx"); >> console_log("info","Waiting for answer!\n"); >> new_session.waitForAnswer(10000); >> console_log("info","***ANSWERED\n"); >> do { >> console_log("info","***Playing file . . . \n"); >> playFile("we_are_trying_to_reach.wav"); >> } while(1); >> >> When I run this code, the call is answered and it starts playing >> the file >> before my phone even starts ringing. >> I'm not sure if it's a bug with waitForAnswer, or a configuration >> issue on >> my side. >> >> For the SIP provider, I'm using Gizmo Phone with the following >> settings in >> sip_profiles/outbound/gizmo.xml: >> >> <include> >> <gateway name="gizmo"> >> <param name="username" value="1747xxxxxxx"/> >> <param name="password" value="xxxxxxx"/> >> <param name="realm" value="proxy01.sipphone.com"/> >> <param name="proxy" value="proxy01.sipphone.com"/> >> <param name="register" value="true"/> >> </gateway> >> </include> >> >> I'd really appreciate any help. >> >> Thanks, >> Marc >> >> >> ________________________________ >> >> _______________________________________________ >> 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 >> >> _______________________________________________ >> 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 >> >> > > _______________________________________________ > 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 _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL: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

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luasql
by Luis Jimenez 10 May '08

10 May '08

Hi i installed lua and luasql, and i did test from cli with access to mysql, it works fine. But when i run the script from FS i get the following erro: 2008-05-09 21:03:01 [ERR] mod_lua.cpp:169 lua_parse_and_execute() /opt/freeswitch/scripts/set_called_group.lua:1: module 'luasql.mysql' not found: no field package.preload['luasql.mysql'] no file './luasql/mysql.lua' no file '/usr/local/share/lua/5.1/luasql/mysql.lua' no file '/usr/local/share/lua/5.1/luasql/mysql/init.lua' no file '/usr/local/lib/lua/5.1/luasql/mysql.lua' no file '/usr/local/lib/lua/5.1/luasql/mysql/init.lua' no file './luasql/mysql.so' no file '/usr/local/lib/lua/5.1/luasql/mysql.so' no file '/usr/local/lib/lua/5.1/loadall.so' no file './luasql.so' no file '/usr/local/lib/lua/5.1/luasql.so' no file '/usr/local/lib/lua/5.1/loadall.so' stack traceback: [C]: in function 'require' /opt/freeswitch/scripts/set_called_group.lua:1: in main chunk

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waitForAnswer returns too soon
by Marc Orenberg 09 May '08

09 May '08

Hello everyone. I'm a new FreeSwitch user, and I'm having a problem with waitForAnswer() with SpiderMonkey. I'm using it to make an outbound call as follows: new_session = new Session(); new_session.originate(session, "sofia/gateway/gizmo/1xxxxxxxxxx"); console_log("info","Waiting for answer!\n"); new_session.waitForAnswer(10000); console_log("info","***ANSWERED\n"); do { console_log("info","***Playing file . . . \n"); playFile("we_are_trying_to_reach.wav"); } while(1); When I run this code, the call is answered and it starts playing the file before my phone even starts ringing. I'm not sure if it's a bug with waitForAnswer, or a configuration issue on my side. For the SIP provider, I'm using Gizmo Phone with the following settings in sip_profiles/outbound/gizmo.xml: <include> <gateway name="gizmo"> <param name="username" value="1747xxxxxxx"/> <param name="password" value="xxxxxxx"/> <param name="realm" value="proxy01.sipphone.com"/> <param name="proxy" value="proxy01.sipphone.com"/> <param name="register" value="true"/> </gateway> </include> I'd really appreciate any help. Thanks, Marc

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Japanese voicemail?
by Robert La Ferla 08 May '08

08 May '08

Is there support yet for Japanese sound files and grammar in FS voicemail like Asterisk? Other languages?

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