October 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

voicemail email not working.
by covici＠ccs.covici.com 01 Oct '10

01 Oct '10

Hi. I left a test message and instead of using the variable I have in my user directory, it tried to deliver to username@domain name which is invalid -- so why did it not use the variable voicemail_email I have in the user directory and it completely ignored the voicemail_notify_email I have as well? Any assistance would be appreciated. -- 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

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Simultaneous ring of all phones registered with the same extension number
by Vitalii Colosov 01 Oct '10

01 Oct '10

Hi, Could not find this answer, so would appreciate any hints. I have registered 2 sip phones using the *same *extension number. When I dial this extension number, only one sip phone rings. However I need both of them to start ringing. Is it possible, that when I dial the number - both other phones will ring, not only one..? And when I answer on a phone, another one would stop ringing. Thank you, Vitalie

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Echo on sip trunk
by Alan Dawson 01 Oct '10

01 Oct '10

Hi, I have a freeswitch box, with a sip trunk to a voip provider. When making / receiving external calls the external party gets an echo, which the freeswitch users dont suffer from. What can I do to reduce that. Thanks Alan Dawson -- GPG key: http://aland.burngreave.net/files/e81a4bba.gpg.pub.asc Key Transition: http://aland.burngreave.net/files/keytransition.txt.asc Further Reading: https://we.riseup.net/alster/openpgp-dsa1-key-rollover

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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

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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

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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

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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

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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.

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ip-change detection
by Jonas Gauffin 01 Oct '10

01 Oct '10

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

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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

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