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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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deflect and redirect
by Jonas Gauffin 08 May '08

08 May '08

Hello I don't really know what "deflect" means, but it works as redirect on established calls. Some of you might use redirect in your dialplans, and it will not work on established calls (for instance if you transfer a call in a javascript using the dialplan). In this case you need to use the deflect app (which doesn't work with new calls). This means that your dial plan need to use redirect on new calls and deflect on established calls. I solved it by setting a session variable everything I do a transfer and then check it to determine if redirect/deflect should be used. //Jonas

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WaitForSilence
by Nick Temple 08 May '08

08 May '08

Is there an API or module that implements a routine similar to Asterisk's "WaitForSilence"? I'm looking for very basic "wait until someone is done speaking before playing audio" functionality. Or maybe an example app that does something similar? I've searched the site, but don't see anything immediately relevant. Thanks, Nick P.S: Freeswitch is working extremely well. I'll have a great testimonial and maybe case study for you once the current project is complete - right around 1st of June (days after 1.0 is released, hopefully)

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Cepstral 5 support
by Brendan Gilmore 08 May '08

08 May '08

Is there anyway to use Cepstral version 5 voices? I successfully setup mod_cepstral according to the wiki. I then purchased a voice to use, but discovered they are selling version 5 now. I rebuilt Freeswitch and mod_cepstral but getting the **libswift.so.4: cannot open shared object file: No such file or directory** error. Since there is no longer a libswift.so.4 file, what do I need to do to get it to work? -- Brendan

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