April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS to answer a call with ZRTP and do text-to-speech of SaS
by Fabio Pietrosanti (naif) 01 Apr '10

01 Apr '10

Hi all, i would like to call from the product my company is going to release (www.privatewave.com, mobile voice encryption with ZRTP for Nokia, iPhone, Blackberry, Android) a FS extension and, without the ZRTP enrollment, having the extension to be answered negotiating ZRTP and playing with Text to Speech the SaS . Is this possible with current FS/ZRTP integration? Fabio

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Trying to test mod_silk
by Jason White 01 Apr '10

01 Apr '10

I am trying to test mod_silk with FreeSWITCH rev. 17118. The problem appears to be that I haven't been able to get my FreeSWITCH to offer SILK as one of its codecs. mod_silk is loaded. I am using the PortAudio interface to make the call; the other end is running a recently built FreeSWITCH as well, thanks to Bill Acker for participating in the testing as always. I have global_codec_prefs and outbound_codec_prefs in vars.xml configured with SILK@24000h as the first item in the comma-separated list. I have also tried it with SILK (no sampling rate specified). The SIP trace doesn't show that it is being offered. Bill has configured the other end in the same way. For example: a=rtpmap:10 L16/48000 a=rtpmap:114 CELT/48000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 is what appears as the SIP message from my local FreeSWITCH with SILK@24000h,CELT@48000h,G722,BV32,PCMU,PCMA,GSM in the codec preferences. (Actually, bv32 is out of date; I know the name changed and I should fix it - I haven't been playing with these settings for a while.) Full SIP traces are available if desired. It's x86_64 Debian GNU/Linux and IPv6 for making the connection. I know Brian and others have this working, so there's obviously something amiss here. As usual, there's no urgency - it's another attempt at a community contribution through testing.

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ZRTP on the FreeSWITCH PBX - questions on the log report
by SERGE TUMBA 01 Apr '10

01 Apr '10

I recently deployed ZRTP on the FreeSWITCH PBX on a Linux machine and I have two X-Lite softphones (on Windows) properly connected and secure with zfone and I am wondering why the FreeSWITCH log is returning the following errors. Also, the zfone on the machine receiving calls won't stay secure for a longtime after the two softphones are connected. I need help to understand this. Thank you! 2010-03-28 21:40:11.870118 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:11.890334 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:11.910739 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:11.930733 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:11.950166 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:11.970874 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:11.990705 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:12.128549 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:12.130517 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:12.150576 [ERR] switch_rtp.c:2848 Error: zRTP protection drop with code 9 2010-03-28 21:40:12.170183 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:12.190531 [ERR] switch_rtp.c:2848 Error: zRTP protection drop with code 9 2010-03-28 21:40:12.210543 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:13.799105 [NOTICE] sofia.c:4789 Hangup sofia/internal/1000(a)192.168.118.178 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2010-03-28 21:40:13.811844 [NOTICE] switch_ivr_originate.c:3079 Hangup sofia/internal/sip:1001@192.168.118.128:12430 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2010-03-28 21:40:13.811844 [ERR] switch_ivr_originate.c:2430 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2010-03-28 21:40:13.811844 [INFO] mod_dptools.c:2353 Originate Failed. Cause: ORIGINATOR_CANCEL 2010-03-28 21:40:13.863496 [NOTICE] switch_core_session.c:1179 Session 17 (sofia/internal/1000(a)192.168.118.178) Ended 2010-03-28 21:40:13.863496 [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/1000(a)192.168.118.178 [CS_DESTROY] 2010-03-28 21:40:13.935508 [NOTICE] switch_core_session.c:1179 Session 18 (sofia/internal/sip:1001@192.168.118.128:12430) Ended 2010-03-28 21:40:13.935508 [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/sip:1001@192.168.118.128:12430 [CS_DESTROY] 2010-03-28 21:40:35.542292 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1001(a)192.168.118.178 [d34e988a-82b4-4891-8c19-82a695eb8cc2] 2010-03-28 21:40:35.562382 [INFO] mod_dialplan_xml.c:418 Processing 1001->1000 in context default 2010-03-28 21:40:35.562382 [INFO] switch_ivr_async.c:2339 Bound B-Leg: 1 execute_extension::dx XML features 2010-03-28 21:40:35.562382 [INFO] switch_ivr_async.c:2339 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/1001.2010-03-28-21-40-35.wav 2010-03-28 21:40:35.562382 [INFO] switch_ivr_async.c:2339 Bound B-Leg: 3 execute_extension::cf XML features 2010-03-28 21:40:35.562382 [NOTICE] switch_channel.c:669 New Channel sofia/internal/sip:1000@192.168.118.179:22558 [bd9f7fa3-426a-4cbe-8a67-9c531d14bba0] 2010-03-28 21:40:36.064353 [INFO] sofia.c:662 Update Callee ID to "1000" <1000> 2010-03-28 21:40:36.085286 [NOTICE] sofia.c:4223 Ring-Ready sofia/internal/sip:1000@192.168.118.179:22558! 2010-03-28 21:40:36.138043 [INFO] switch_ivr_originate.c:1064 Sending early media 2010-03-28 21:40:36.139955 [NOTICE] mod_sofia.c:1907 Pre-Answer sofia/internal/1001(a)192.168.118.178! 2010-03-28 21:40:36.310490 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:36.423250 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:36.430517 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:36.451432 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:36.574454 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:36.590511 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:36.610380 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:56.354321 [NOTICE] sofia.c:4733 Channel [sofia/internal/sip:1000@192.168.118.179:22558] has been answered 2010-03-28 21:40:56.410969 [NOTICE] switch_ivr_originate.c:3103 Channel [sofia/internal/1001(a)192.168.118.178] has been answered 2010-03-28 21:40:56.510922 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 2010-03-28 21:40:56.530552 [ERR] switch_rtp.c:2848 Error: zRTP protection drop with code 9 2010-03-28 21:40:56.551849 [ERR] switch_rtp.c:2848 Error: zRTP protection drop with code 9 2010-03-28 21:40:56.619175 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9 _________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/210850553/direct/01/

2 1

mod_unimrcp with Nuance "Timed out waiting for channel to be ready"
by Adam Meehan 01 Apr '10

01 Apr '10

I'm using mod_unimrcp module with the Nuance Speech Server 5.1 and I'm getting the "Timed out waiting for channel to be ready" when trying to use TTS. This is using MRCP v2 and FS trunk rev 17126. I've not used MRCP and I am stuck. Any suggestions? Thanks Adam

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