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my info as appearing in conference
by covici＠ccs.covici.com 01 Oct '10

01 Oct '10

Hi. If I use portaudio to go into the freeswitch conference, I get allzeroes for the number and something like test for the name -- whereas if I go into an asterisk conference, it gives me the correct information using the caller id number and name. So are there some other variables I need to set or what else is going on here? Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

3 7

Duplicate CDRs
by Saeed Ahmed 01 Oct '10

01 Oct '10

Dear List, I am using xml_cdr to write the records on disk. today for first time when there was a call pressure around 60-70 concurrent calls (around 140 session), then FS started to write same CDR record two times with different/unique file name and also different/unique UUIDs etc.. but the called number, start time, end time everything is same. so its surely a duplicated CDR. -> I am running FS on cent os, on a 6 core machine with 4 GM RAM. -> FS version: FreeSWITCH Version 1.0.head (git-72baaf6 2010-09-21 19-22-28 -0500) (is anything regarding cdr changed in new version? so i can upgrade) > Please note that system was running fine with few call attempts and i never faced this prob, only in case of 6-70 councurrent calls My xml_cr.conf: <configuration name="xml_cdr.conf" description="XML CDR CURL logger"> <settings>      * <param name="retries" value="2"/>*       * <param name="log-dir" value="/opt/fscdr"/>*      * <param name="prefix-a-leg" value="true"/>*  *<param name="encode" value="true"/>*     *<param name="err-log-dir" value="/opt/error_cdr"/>*                 </settings> </configuration> Many Thanks - Saeed

3 3

License of libzrtp for usage with FreeSWITCH?
by Patrick Lists 01 Oct '10

01 Oct '10

Hi, Does anyone know what the license is of libzrtp for usage with FreeSWITCH? In the FreeSWITCH source I see a file docs/zrtp_agpl-3.0.txt but the libzrtp source itself is not included. So I got libzrtp-0.81.514 and in the source there is a file zrtp_legal.c that says: * libZRTP SDK library, implements the ZRTP secure VoIP protocol. * Copyright (c) 2006-2009 Philip R. Zimmermann. All rights reserved. * * The version of the SDK library that is packaged with this legal notice is * NOT licensed under the GPL, AGPL, LGPL, or any other open source license. * For licensing terms or other information, * contact Philip R. Zimmermann <prz(a)mit.edu>. * For more contact information, see http://philzimmermann.com * * This file must be packaged together with the rest of the libZRTP SDK * source code. That's why it's in a .c file. * * This software may be subject to export controls by the US Commerce * Department's Bureau of Industry and Security. This software is provided * "as is," with no warranty expressed or implied. This is quite the opposite of what zrtp_agpl-3.0.txt suggests in the FreeSWITCH source. Any ideas? Regards, Patrick

4 6

serious problem in record_session
by covici＠ccs.covici.com 01 Oct '10

01 Oct '10

Hi. I am having a serious problem making record_session work properly in my dialplan -- it records fine using meta-app, but if I turn it off and on, it overwrites the previous file with the same time and date when the meta-app was invoked. Is there anyway to make the time the current time so it would give another file? Thanks for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

5 12

PRI inbound problem after updating freeswitch to current git version
by ovvenkat 01 Oct '10

01 Oct '10

Hi to all, I have upgraded my freeSwitch to current git version. I am using below dial plan to handle the inbound call (SangomaA101 AND openzap) <extension name="start_Talk"> <condition field="destination_number" expression="^1100$"> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> When I am calling the PRI number, my phone is never answered, its keep on ringing. But, fs_cli logs shows that, demo_ivr menu is playing. here is my fs_cli output, http://pastebin.freeswitch.org/14098 Its happening after upgrading the freeswitch to git version. Any one plz help me what may cause this problem? Thanks. -- Regards Venkatesan OV.

1 1

ip-change detection
by Jonas Gauffin 01 Oct '10

01 Oct '10

Hello, How do I turn off ip-change detection? Regards, Jonas

2 2

Error forcing iLBC to 30 ms
by Juan Antonio Ibañez Santorum 01 Oct '10

01 Oct '10

Hello, I am trying to use wiki config about iLBC codec but I am not be able to make it work. If I configure codec prefs as: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA,iLBC"/> I get: ------------------------- 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec Compare [iLBC:97:8000:30]/[GSM:3:8000:20] 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec Compare [iLBC:97:8000:30]/[PCMU:0:8000:20] 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec Compare [iLBC:97:8000:30]/[PCMA:8:8000:20] 2010-10-01 08:48:22.446360 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/dsp1(a)174.36.98.82) Running State Change CS_NEW 2010-10-01 08:48:22.446360 [DEBUG] sofia_glue.c:3867 Audio Codec Compare [iLBC:97:8000:30]/[iLBC:98:8000:20] ------------------------- Phone offers 30 ms, FS 20 ms and call gets cursed... If I use: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA,iLBC@30i"/> or <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA,iLBC@20i"/> FS does not offer iLBC codec in Audio Codec Compare Where could be the misscounfiguration? Which would be the best config? Regards

3 3

Not able to connect mysql using Dbh
by ovvenkat 01 Oct '10

01 Oct '10

Hi to all, I could not able to connect to mySql using Dbh. Here is the coding I have used to connect to my db. * local dbh = assert(freeswitch.Dbh("dsn_name","venkat","venkat123"))* I am getting following error while executing the lua test.lua in fs_cli 2010-09-29 17:57:29.924498 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/test.lua:23: attempt to call field 'Dbh' (a nil value) stack traceback: /usr/local/freeswitch/scripts/test.lua:23: in main chunk */etc/odbc.ini* file contains [dsn_name] Driver=MySQL SERVER=localhost PORT=3306 USER=venkat PASSWORD=venkat123 DATABASE=venkat Socket = /var/lib/mysql/mysql.sock *I can able to connect to my db using following command * isql dsn_name venkat venkat123 I trying to figure it out for past two days and gone though wiki not able to crack it. Any one please help me to resolve my issue. -- Regards Venkatesan OV.

4 7

FW: C# Examples
by Tim B 01 Oct '10

01 Oct '10

Does anyone have any real world C# examples they can share? I am looking how to use C# to handle freeswitch configuration, and call control. Any help would be appreciated. Tim

2 1

Codec Mismatch Because of Different Frame Size
by Jan Riedinger 30 Sep '10

30 Sep '10

I'm using the following configuration: vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA,G723,G7221@32000h,G7221@16000h,G722,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,G723,GSM"/> external.xml (my standard sip profile): <param name="inbound-codec-negotiation" value="greedy"/> <param name="inbound-late-negotiation" value="false"/> <param name="disable-transcoding" value="true"/> With this configuration I want to restrict the allowed codecs, force the usage of G729 as first choice, if available and to minimize the calls with transcoding. I don't want to force the usage of a specific frame size. Now a lot of calls get rejected with "incompatible destination" by FreeSwitch, because the frame size of the incoming calls differs from the Freeswitch default (mostly 20 ms), see the debug log below. Is this the intended behaviour? If it is, how can I circumvent it? Do I have to list every codec with every possible frame size in the global_codec_prefs list? Thank you in advance Jan [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia.c:4469 Channel sofia/external/0049xxx(a)X.X.X.X entering state [received][100] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia.c:4480 Remote SDP: v=0 o=- 554229 1 IN IP4 X.X.X.X s=- c=IN IP4 X.X.X.X t=0 0 a=sendrecv m=audio 26044 RTP/AVP 18 4 0 8 96 101 a=fmtp:18 annexb=yes a=rtpmap:96 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=ptime:60 a=silenceSupp:off - - - - [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G729:18:8000:60:0]/[G729:18:8000:20:8000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G723:4:8000:60:0]/[G729:18:8000:20:8000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMU:0:8000:60:0]/[G729:18:8000:20:8000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMA:8:8000:60:0]/[G729:18:8000:20:8000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G726-32:96:8000:60:0]/[G729:18:8000:20:8000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [telephone-event:101:8000:60:0]/[G729:18:8000:20:8000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4335 Set 2833 dtmf send/recv payload to 101 [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G729:18:8000:60:0]/[PCMU:0:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G723:4:8000:60:0]/[PCMU:0:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMU:0:8000:60:0]/[PCMU:0:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMA:8:8000:60:0]/[PCMU:0:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G726-32:96:8000:60:0]/[PCMU:0:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [telephone-event:101:8000:60:0]/[PCMU:0:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4335 Set 2833 dtmf send/recv payload to 101 [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G729:18:8000:60:0]/[PCMA:8:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G723:4:8000:60:0]/[PCMA:8:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMU:0:8000:60:0]/[PCMA:8:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMA:8:8000:60:0]/[PCMA:8:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G726-32:96:8000:60:0]/[PCMA:8:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [telephone-event:101:8000:60:0]/[PCMA:8:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4335 Set 2833 dtmf send/recv payload to 101 [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G729:18:8000:60:0]/[G7221:115:32000:20:48000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G723:4:8000:60:0]/[G7221:115:32000:20:48000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMU:0:8000:60:0]/[G7221:115:32000:20:48000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMA:8:8000:60:0]/[G7221:115:32000:20:48000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G726-32:96:8000:60:0]/[G7221:115:32000:20:48000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [telephone-event:101:8000:60:0]/[G7221:115:32000:20:48000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4335 Set 2833 dtmf send/recv payload to 101 [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G729:18:8000:60:0]/[G7221:107:16000:20:32000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G723:4:8000:60:0]/[G7221:107:16000:20:32000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMU:0:8000:60:0]/[G7221:107:16000:20:32000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMA:8:8000:60:0]/[G7221:107:16000:20:32000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G726-32:96:8000:60:0]/[G7221:107:16000:20:32000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [telephone-event:101:8000:60:0]/[G7221:107:16000:20:32000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4335 Set 2833 dtmf send/recv payload to 101 [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G729:18:8000:60:0]/[G722:9:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G723:4:8000:60:0]/[G722:9:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMU:0:8000:60:0]/[G722:9:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMA:8:8000:60:0]/[G722:9:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G726-32:96:8000:60:0]/[G722:9:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [telephone-event:101:8000:60:0]/[G722:9:8000:20:64000] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4335 Set 2833 dtmf send/recv payload to 101 [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G729:18:8000:60:0]/[GSM:3:8000:20:13200] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G723:4:8000:60:0]/[GSM:3:8000:20:13200] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMU:0:8000:60:0]/[GSM:3:8000:20:13200] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [PCMA:8:8000:60:0]/[GSM:3:8000:20:13200] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [G726-32:96:8000:60:0]/[GSM:3:8000:20:13200] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4233 Audio Codec Compare [telephone-event:101:8000:60:0]/[GSM:3:8000:20:13200] [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4335 Set 2833 dtmf send/recv payload to 101 [m[33m2010-09-30 10:25:45.798068 [DEBUG] sofia_glue.c:4335 Set 2833 dtmf send/recv payload to 101 [m[33m2010-09-30 10:25:45.799052 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0049xxx(a)x.x.x.x) Running State Change CS_NEW [m[33m2010-09-30 10:25:45.799052 [DEBUG] switch_core_state_machine.c:320 (sofia/external/0049xxx(a)x.x.x.x) State NEW [m[33m2010-09-30 10:25:45.799052 [DEBUG] switch_channel.c:2359 (sofia/external/0049xxx(a)x.x.x.x) Callstate Change DOWN -> HANGUP [m[33m2010-09-30 10:25:45.799052 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0049xxx(a)x.x.x.x) Running State Change CS_HANGUP [m[36m2010-09-30 10:25:45.799052 [NOTICE] sofia.c:4686 Hangup sofia/external/0049xxx(a)x.x.x.x [CS_NEW] [INCOMPATIBLE_DESTINATION] -- Jan Riedinger Phone : +49-30-39 73 19 66 Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64 E-Mail: riedinger(a)sns.eu SNS Consult GmbH ICQ : 163-237-041 Südwestkorso 49a MSN : jan(a)sns-consult.de 14197 Berlin GERMANY Skype : Jan Riedinger AG Charlottenburg - HRB 71973

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