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git HEAD 503 Maximum Calls In Progress
by Seven Du 10 Feb '11

10 Feb '11

Hi, It's a fresh install with wget freeswitch.org/eg/Makefile && make, and I'm sure I successfully called 9196 and everything was ok. But a few minutes later I started getting 503 Maximum Calls In Progress also originate user/1003 or loopback/9196 shows errors see http://pastebin.freeswitch.org/15333 It's on a ubuntu 8.04 32bit, the only thing I did between non-work and work was I configured odbc with postgresql to work with isql. I haven't touch the FS config yet. I don't think they are related. here is also a nua debug http://pastebin.freeswitch.org/15335 Any hint on this? Thanks. btw, I also noticed there might be some network problems between me and the server where sometime sip packets not shown ( sofia debug and ngrep) on server. Anyway, since loopback cannot create sessions so something must be wrong in FS side, but what's that? Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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Re: [Freeswitch-users] How to read RTP package?
by Frankie Yiu 10 Feb '11

10 Feb '11

Here is the description of the project: We want to build a system that would play a recorded message when a human or an answer machine picks up the phone(eventually for multiple of phone numbers). For a human, we want to play the message after the human speaks. For answer machine, we want to wait for the beep. To do this, we think that we have to get the RTP packets and determine when is the right timing to play the message. Currently, we have code in C# to play a recorded message when someone picks up the phone. Someone suggests me to try Media bugs but it is not working with the wrapper function for C# (swig code, because of the pointer to pointer type, unless I am wrong). So as our test, we call a wrapper function from C# with a "session" information, and inside the C++ code, we initialize a media bug with a call back function for READ. I got event for Media Bugs INIT and when Media Bugs CLOSE message but never got any READ event. Here are my questions: 1) How can I get the READ event from Media bugs event? It seems to me that when I call a routine to initialize(ADD) the Media bugs, I will get INIT and CLOSE event back right away event though we have not hung up the phone yet, and therefore we never got any READ events. 2) If I am calling a number from C# code, how can I pass the session information to C++ code? (as I mention earlier I have modified function inside freeswitch_wrap.cxx to have call back in C++ code, but this is not the proper way) 3) Is there a way to have packet send back to C# (I think those type "SWIGTYPE_p_p_..." is not working properly for C#, am I correct?) Thanks in advance. Any help would be great!! Frankie

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C# problem for calling "freeswitch.switch_core_session_read_frame()"
by Frankie Yiu 10 Feb '11

10 Feb '11

Hi there, I would like to dial a phone number, and read the RTP package back when connected so that we can analyze the data; (to determine when we should play an audio at the right time--human or answer machine.) We found a code for IVR test (http://docs.freeswitch.org/switch__ivr_8c-source.html) Function name: switch_ivr_sound_test We would like to do the same but with C# code, but we encountered a problem when calling the function "freeswitch.switch_core_session_read_frame( ) " < in swig.cs switch_core_session_read_frame(SWIGTYPE_p_switch_core_session session, SWIGTYPE_p_p_switch_frame frame, uint flags, int stream_id) > where the second parameter--frame is a pointer to pointer of switch_frame type and in C# code, it is having a difficulty passing an object to the C++ side and keep the pointer place holder before going deeper into the C code (switch_core_io.c) 1) So is there any way I can call this function in C#? 2) Is there another function or routine that you can suggest me to for reading RTP package? Thanks, Frank

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Polycom and registering with domains (user@domain.tld@domain.tld)
by Gabriel Gunderson 10 Feb '11

10 Feb '11

All, I have a multi-tenant FreeSWITCH server up and running. We have all the proper SVR records setup for each domain and everything works great. Most SIP clients Just Work (tm), however, the new Polycom phones I have don't seem to auth properly with domains. I've tried every combination of settings (via the web config, I haven't setup a TFTP server yet). The issue is that auth has the domain twice, like so: user@domain.tld@domain.tld. It seems like the Internet has threads of discussion where others express frustration with this, but I can't seem to find a documented solution. It's maddening because every other User Agent Client works swimmingly and no matter what I've tried, I can't get the right combination of settings on the Polycom. I love the phones, but this is getting old :) Anyway, any pointers for me? Thanks, Gabe sofia.conf: ************************************************************* [SNIP] <domains> <domain name="all" alias="true" parse="false" /> </domains> <settings> <param name="log-auth-failures" value="true" /> <param name="forward-unsolicited-mwi-notify" value="false" /> <param name="context" value="public" /> <param name="sip-port" value="$${internal_sip_port}" /> <param name="dialplan" value="XML" /> <param name="inbound-codec-prefs" value="$${global_codec_prefs}" /> <param name="outbound-codec-prefs" value="$${global_codec_prefs}" /> <param name="rtp-ip" value="$${internal_ip}" /> <param name="sip-ip" value="$${internal_ip}" /> <param name="hold-music" value="$${hold_music}" /> <param name="record-path" value="$${recordings_dir}" /> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav" /> <param name="auth-calls" value="true" /> <param name="ext-rtp-ip" value="auto-nat" /> <param name="ext-sip-ip" value="auto-nat" /> </settings> [/SNIP]

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skypopen: problem loading the custom alsa driver
by Giovanni Maruzzelli 10 Feb '11

10 Feb '11

Thomas wrote: ================= Hi Giovanni, thankx for you replay. I using ubuntu lucid. The alsa-driver make a problem. Alsa is compiled correcly from sources. when i execute the following command: root@freeswitch:/# modprobe snd_dummy WARNING: Error inserting snd_page_alloc (/lib/modules/2.6.32-27-generic-pae/kern el/sound/acore/snd-page-alloc.ko): Unknown symbol in module, or unknown paramete r (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.32-27-generic-pae/kernel/so und/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dme sg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.32-27-generic-pae/kernel/soun d/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_dummy (/lib/modules/2.6.32-27-generic-pae/kernel/soun d/drivers/snd-dummy.ko): Unknown symbol in module, or unknown parameter (see dme sg) i make: depmod -a i reboot the system, the problem is the same. can you looking please? tahkx. ===================== My answer: have you followed strictly the guidelines in the wiki page for compiling and loading the custom snd-dummy? (alsa and drivers are tricky beasts) can you paste here the results in dmesg (/var/log/messages) that are related to that? anyway, seems that you have one of this problems: a) maybe you have not done "make" then (after make) "make install" b) maybe you still have the old sound directory in your /var/lib/modules tree c) maybe you have compiled and installed the alsa mods for a previous kernel version can you report exactly what you have done (and answer the previous questions) ? -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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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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