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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 (Actually, bv32 is out of date; I know the name changed and I should fix it - I haven't been playing with these settings for a while.)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/

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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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Call Recording & bind_meta_app
by Bradley Brashier 31 Mar '10

31 Mar '10

I need to add the ability to record a call to an existing infrastructure, and I'm intending to use FreeSwitch to do this. The recording needs to happen upon a DTMF command, which I understand is normally done via bind_meta_app. From what I can tell, bind_meta_app requires that you use the * key in addition to some other key. Unfortunately, the system is already doing something else with *, so that's not a key I can use to get this to work. My question, then: what is the best way to record a call using DTMFs that do not include *? - Should I have Freeswitch forward an appropriate DTMF to a controlling application that starts recording? - Should I try to modify the bind_meta_app code to use another digit instead (say, #)? If so, where is the right place to look for this? As a separate, but related question, I also need to play a disclaimer when the recording starts ("This call is now being recorded."). Is it OK to use bind_meta_app to bind 2 apps to the same DTMF combo? BB

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ClueCon MMX - It Will Be Here Before You Know It!
by Michael Collins 31 Mar '10

31 Mar '10

Greetings! We wanted to remind everyone to get a head start on their ClueCon arrangements. We have great things planned for this year and we don't want anyone to miss out. Please call 1-877-742-CLUE (2583) to register or if you have any questions at all about ClueCon. See you in August! -ClueCon Team http://www.cluecon.com 877.742.CLUE

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Phone Registration Events
by Dave Stevenson 31 Mar '10

31 Mar '10

Hi, Is there a log option to turn on logging of SIP registration messages ? I have a mixture of phones and am having a problem with some of them periodically losing registration with FreeSWITCH. All the phones start off working after provisioning and have registration expiry times configured of 3600 seconds. I have tried changing the expiry time to no effect and I wanted to see if the phones were actually trying to renew the registration when they should. Regards Dave

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SILK Codec
by Brian West 31 Mar '10

31 Mar '10

We now have mod_silk in tree it has only been tested on Linux so far. The silk library itself was released in the IETF draft only an hour and twenty minutes ago. I wrote a perl script to extract the source from the draft. Checked in the lib and the codec module which I had written to the binary lib I had a few months ago. It has more work to do... but its there if anyone wants to chip in and libtoolize the library, revamp the build system and assist in testing the codec on multiple platforms. Thanks, Brian

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SIP Call Trace
by Ken Fulmer 31 Mar '10

31 Mar '10

Perhaps I've missed it, but can someone explain how to capture SIP and SDP messages for troubleshooting? Thanks, Ken Fulmer

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Adding Record-Route Header
by Suneel Papineni 31 Mar '10

31 Mar '10

Hi, How can I add Record-Route (pointing to another FS acting as Proxy server) for the calls handled by to FSComm client. Also can I set this while making a call or accepting a call using event socket commands (pa answer / pa call xxxx). Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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bridge from-user
by Justin B Newman 31 Mar '10

31 Mar '10

I have tried a number of variations on: <action application="bridge" data="[from-user=abc,sip_auth_username=abc,sip_auth_password=123]sofia/external/$1(a)voip.mydomain.com"/> So far, to no avail. In all cases, the From does not contain "abc", but rather, the Caller-ID data. FreeSWITCH Version 1.0.5-20100329-0400 (17128M) Is it possible to modify From when placing an outbound call in absence of a gateway? -jbn

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