February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

git-a93623 version (2011/02/07) causes [DESTINATION_OUT_OF_ORDER]
by mazilo 10 Feb '11

10 Feb '11

This morning, when I did a git pull on my local FS git root directory, I noticed there is a significant update (including the main switch and/or ivr). So, I went a head to compile/install/run this git version to find out the following error message through fs_cli when I placed an outbound call: FreeSWITCH Version 1.0.head (git-a936236 2011-02-07 16-33-45 -0600) 2011-02-08 11:42:05.752459 [CRIT] mod_sofia.c:3844 Error Creating Session 2011-02-08 11:42:05.752459 [ERR] switch_ivr_originate.c:2638 Cannot create outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER] The dialplan context works with previous git version. So, do I need to make changes to my dialplan context? ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/git-a93623-version-2011-02-07… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] ringback settings
by Madovsky 10 Feb '11

10 Feb '11

found the problem, before bridge I put <action application="set_audio_level" data="read 1"/> <action application="set_audio_level" data="write 1"/> FS doesn't like it. I guess it needs the bridge done to run them Any ? THanks ----- Original Message ----- From: "Madovsky" <infos(a)madovsky.org> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Wednesday, February 09, 2011 8:32 PM Subject: Re: [Freeswitch-users] ringback settings > So what the solution ? > > > ----- Original Message ----- > From: "curriegrad2004" <curriegrad2004(a)gmail.com> > To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> > Sent: Wednesday, February 09, 2011 8:13 PM > Subject: Re: [Freeswitch-users] ringback settings > > >> Maybe your provider is blocking the ringbacks set by you? >> >> On Wed, Feb 9, 2011 at 4:42 PM, Madovsky <infos(a)madovsky.org> wrote: >>> I don't find the right settings >>> to get ringback from a mobile phone and landline >>> when I call to a DID< >>> I put instant_ringback=true and/or application ring_ready >>> but no success >>> >>> thanks >>> _______________________________________________ >>> 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

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ringback settings
by Madovsky 10 Feb '11

10 Feb '11

I don't find the right settings to get ringback from a mobile phone and landline when I call to a DID< I put instant_ringback=true and/or application ring_ready but no success thanks

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Confusing SIP auth failure logging message?
by Simon J Mudd 10 Feb '11

10 Feb '11

I've been looking at trying to configure tighter controls for extensions that register. Doing so made me trigger this error message (adjusted slightly): 2011-02-07 00:07:51.343303 [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000(a)sip.example.com] from ip 192.168.4.99 2011-02-07 00:07:51.343303 [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000(a)sip.example.com] from ip 192.168.4.99 Looking at the message it is not clear if the SIP authentication has succeeded or failed. Judging by the code it seems this is meant to represent a SIP auth failure. If so should the code not be patched as shown? diff --git a/src/mod/endpoints/mod_sofia/sofia_reg.c b/src/mod/endpoints/mod_sofia/sofia_reg.c index 631cbdb..e42c777 100644 --- a/src/mod/endpoints/mod_sofia/sofia_reg.c +++ b/src/mod/endpoints/mod_sofia/sofia_reg.c @@ -1244,7 +1244,7 @@ uint8_t sofia_reg_handle_register(nua_t *nua, sofia_profile_t *profile, nua_hand } /* Log line added to support Fail2Ban */ if (sofia_test_pflag(profile, PFLAG_LOG_AUTH_FAIL)) { - switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "SIP auth challenge (%s) on sofia profile '%s' " + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "SIP auth failure (%s) on sofia profile '%s' " "for [%s@%s] from ip %s\n", (regtype == REG_INVITE) ? "INVITE" : "REGISTER", profile->name, to_user, to_host, network_ip); } Thanks, Simon

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Best way to implement real-time ivr from mysql
by Tim St. Pierre 09 Feb '11

09 Feb '11

Hello list, I'm trying to implement a flexible, but easy to administer IVR system for our multi-tenant PBX. My goal is to be able to build a flexible auto-attendent scheme for incoming calls, based on data in a mysql database. The functionality I need is: -Different options based on time of day -Playback of 1-2 audio files while listening for digits (Thank you for calling abc company) -Single digit options set some variables, then transfer to a defined extension (press 1 for sales) -Multi-digit options are checked against a pattern, then a transfer is executed (enter the extension now ...) Not very complicated, I know, but making it scale is tricky. I have tried LUA, but LuaSQL has issues. Even using ODBC, I still get memory leaks and random errors with file handles etc. For the most part, I haven't had any issues with xml_curl getting config from another server that generates it with PHP. I'm exploring this, but it looks like I'll have to implement the following logic: -Call comes in, and a dialplan is returned that plays the greetings, sets some variables, and does a play_and_get_digits, followed by a transfer back to the dialplan, so we can figure out what to do based on the digits. -The callback to the dialplan evaluates the dtmf presented, and decides what action to take, returning that in another dialplan piece. I think this could work. but it means that a dialplan lookup has to be done each time someone makes a selection. Is there a better way, or does that make the most sense? Thanks! -Tim

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Gateway with no registration
by Cliff Wells 09 Feb '11

09 Feb '11

I'd like to setup a gateway with no registration (other end allows by IP). <include> <gateway name="outbound"> <param name="username" value="none"/> <param name="password" value="none"/> <param name="proxy" value="x.x.x.x"/> <param name="register-transport" value="udp"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> </gateway> </include> As I understand it, the username and password shouldn't matter since registration is challenge/auth. What happens is I get a 403 when FreeSWITCH attempts to register (before any calls are placed), so FreeSWITCH has the gateway marked as FAIL_WAIT. Is this proof the other end is requiring registration? Or is FS expecting a challenge, getting none and marking the gateway down? Most of the solutions I've seen for this type of setup involve using a dialplan rather than a gateway, but my dialplans are served via mod_xml_curl and expect a functional gateway named "outbound" on the FS box. I'd rather not clutter up the xml dialplans with a bunch of special cases, so a named gateway is preferred. The problem certainly may lie on the other end (I don't have control of that system), but I'm not sure how to verify that. Regards, Cliff

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RTP keep alive
by Sam 09 Feb '11

09 Feb '11

Hello, We have a situation wherein we need to keep alive RTP ,is there any parameter to do that, because when someone is on long conversation and not talking fro brief duration the call disconnects . Regards Sam

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Re-order or Invalid
by Peder 09 Feb '11

09 Feb '11

What is the correct way to send a re-order or invalid message/tone to a caller? We have a couple hundred DIDs and quite a few of them are not used. If someone calls into one of those, we just want to send a re-order or busy tone. My preference would be to NOT answer the call as I would prefer not to be charged for the call if I can avoid it. Peder

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FreeSWITCH Conference Call Today
by Michael Collins 09 Feb '11

09 Feb '11

Hey gang, We decided that we'd have an open call today. No real agenda items, but we're happy to hang out and maybe answer a few questions. -Michael

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ext-rtp-ip value behind NAT
by mazilo 09 Feb '11

09 Feb '11

With FreeSWITCH Version 1.0.head (git-2944364 2011-02-04 16-53-38 -0600) on my Seagate DockStar, I configured my jingle profile using the following two different options: 1. 2. With option #1, no incoming call and caller got intercepted by GV voicemail. fs_cli dumped this http://pastebin.com/8sJjUZx1 message as seen on line# 26 and here is the http://pastebin.com/gnb8dXmW output of dl_debug on. With option #2, the call comes in. When I picked up to answer the call, no audio and caller still heard the ringing tone until the call got intercepted by GV voicemail. fs_cli dumped this http://pastebin.com/fJ1s14tX message and the http://pastebin.com/mHBFECs3 output of dl_debug on. If anyone here has any idea how to fix this, I sure would appreciate that. Thanks. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ext-rtp-ip-value-behind-NAT-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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