February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Vestec Speech Engine: Windows Release
by Kashif Kahn 10 Feb '11

10 Feb '11

Hello Everyone, Vestec recently launched a Windows version of its latest generation speech recognition engine (ASR 2.1) for use with Freeswitch and would appreciate if people could try it. We already support all major Linux distributions. Our ASR engine is designed for "command-and-control" type IVR applications and offers the best deal around for enabling sophisticated speech recognition with Freeswitch. It can be used to speech enable a wide variety of DTMF menus or business processes via keywords-based speech interaction. A starter kit - comprising a specially priced perpetual license for a full-function, full-feature standard ASR engine - is available for $25. Acoustic models cost an additional $9.99 per language. Please visit Vestec webstore at: http://www.vestec.com/products Regards, -Kashif -- Kashif Kahn VP Business Development Vestec Inc Waterloo, ON Canada phone: +1 519 885-7615

1 0

How to dial a remote_skypename from a telephone keypad?
by mazilo 10 Feb '11

10 Feb '11

If one uses mod_skypopen + skype client on an FS system with and IP Phone and/or a telephone attached to an ATA device as an extension to the FS sytem, how can one place a call to a Skype user whose account isn't in numeric characters? ----- 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/How-to-dial-a-remote-skypenam… Sent from the freeswitch-users mailing list archive at Nabble.com.

4 16

Problem with Outbound ESL in C++
by Stephen Wilde 10 Feb '11

10 Feb '11

I'm trying to write an ESL Outbound application in C++ but I have problem to receive events. My application is: http://pastebin.freeswitch.org/15349 The problem is related to the sending of "event plain ALL" command that receives an error as reply. The same problem if I run the Freeswitch "testserver" application of ESL library: the "event plain" command receives always a reply "-ERR command not found" and also the "linger" command. What I'm missing? Here the log of "testserver" application [INFO] testserver.c:19 mycallback() Connected! 4 [DEBUG] src/esl.c:1141 esl_send() SEND filter unique-id 9e159038-d599-4e18-afe9-a8f4eeffd76b [DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Reply-Text] = [+OK filter added. [unique-id]=[9e159038-d599-4e18-afe9-a8f4eeffd76b]] [DEBUG] src/esl.c:1141 esl_send() SEND event plain SESSION_HEARTBEAT CHANNEL_ANSWER CHANNEL_ORIGINATE CHANNEL_PROGRESS CHANNEL_HANGUP CHANNEL_BRIDGE CHANNEL_UNBRIDGE CHANNEL_OUTGOING CHANNEL_EXECUTE CHANNEL_EXECUTE_COMPLETE DTMF CUSTOM conference::maintenance [DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Reply-Text] = [-ERR command not found] [DEBUG] src/esl.c:1141 esl_send() SEND linger [DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Reply-Text] = [-ERR command not found] [DEBUG] src/esl.c:1141 esl_send() SEND sendmsg call-command: execute execute-app-name: answer [DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] = [text/disconnect-notice] [DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Controlled-Session-UUID] = [9e159038-d599-4e18-afe9-a8f4eeffd76b] [DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Disposition] = [disconnect] [DEBUG] src/esl.c:989 esl_recv_event() RECV HEADER [Content-Length] = [67] [INFO] testserver.c:49 mycallback() Disconnected! 4

1 0

Video on X-lite .. FSV module
by sherrypioro 10 Feb '11

10 Feb '11

Hi, I am trying to record or play video on X-lite through Freeswitch. When I do a video call it says faild to send your video. I have FSV module installed on freeswitch and running. Any Ideas???? Thank you!!!

4 3

contact header
by Sam 10 Feb '11

10 Feb '11

Hi, Is there a way to change the contact header to Contact: <sip:abcd@xxxx:5060 from Contact: <sip:gw+foo@xxxx:5060> Regards Sam

2 1

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

3 6

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

1 0

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

2 1

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]

3 3

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

3 9