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call count bind digit on static conference
by Rodney 17 Nov '11

17 Nov '11

windows server 2008 32bit; freeswitch windows msi 11/4/2011 thank you to everyone who helped me fix my mod_flite tts issue. I went and installed licensed cepstral and loaded mod_cepstral and all is fine now but led me to another issue :) I have followed the instructions at http://wiki.freeswitch.org/wiki/Conference_Announce_Count_Inline with the exception that I did not create a a new"plain" profile i just added the action : <control action="execute_application" digits="8" data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default"/> to the default profile when i go into a static conference room it speaks tts callie and tells me I am the only caller, which is correct but unfortunately if second callers joins they don't get tts and when i press the 8 digit when two or more callers it shows 2011-11-17 15:23:27.284724 {INFO} mod_dptools.c:1316 Conference Conference not found 2011-11-17 15:23:27.305231 {NOTICE} switch_core_session.c:2367 Execute say(en number pronounced $(conf_count)) 2011-11-17 15:23:27.325737 {ERR} mod_say_en.c:130 Parse Error! I have read this (http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075160.h…) but am still stuck, not knowing 1/ how to list correct rooms and 2/ tell my system how to find them using the variable "conference_name" in <action application="set" data="conf_count=${conference ${conference_name}list count}"/> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075160.h… also concerning same issue is that when I do a command "conference list" the output is: Conference conference room 1-127.0.0.1 (1 member rate:8000)13; sofia/external/2064200990@66.54.140.446;b11d94a6-94ff-4955-bc87-13e332a7443;unknown;2064200990;hear:speak:floor;0;0;0;50 I try to "conference conference room 1-127.0.0.1" I get response "conference not found" I am sure that all of these issues are fixed with the same method, I just need help in the right direction. your help is appreciated; thanks.

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File playback delay
by Jason Moran 17 Nov '11

17 Nov '11

I have found that the playback of a file can be delayed to the point that the callee will either hangup (thinking it MUST be over now since it has been quiet for 7-8 seconds) or a Voicemail's silence detection will trigger even though I hadn't finished playing back all of the audio files I had intended to play. Ideas: Does session.stream block (spidermonkey script)? Since I am doing several simultaneous calls if session.stream blocked file access that could account for it (although that would be a terrible calamity if true). Does anybody else have ideas? I have been using pcapsipdump to "record" the calls and I can confirm that the reports are true - in some cases the playback of the next audio file can delay for a long time. I have tested these calls to my mobile phone and office phone without any delays present, but it seems to happen to some people some of the time. Thanks, Jason

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Issue with: answer confirmation + custom ringback + mod_freetdm
by Victor Chukalovskiy 17 Nov '11

17 Nov '11

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Re: [Freeswitch-users] mod_rtmp and a security camera
by Pete Mueller 17 Nov '11

17 Nov '11

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How to Update external IP address on Ubuntu?
by Charles Bujold 17 Nov '11

17 Nov '11

Hi, We are newbies to Ubuntu and Freeswitch. We have Freeswitch installed on Ubuntu with Auto-Nat and it works great. The issue we have is that our ISP only provides Dynamic IP's which do not change that often but when they do everyone goes into panic mode until we remember that by simply re-booting the server everything comes back to normal. My question: Is there a method we can put in place that would automate the external-IP update without having to reboot the server. Can Freeswitch notice that the external IP has change and reset itself without manual intervention? And if we need to reboot the server is there a program that would do this automatically when the external IP changes? Thanks cjb

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Joining a call from Gtalk to a conference on FreeSwitch
by Papineni, Suneel 17 Nov '11

17 Nov '11

Hi, I got the new GIT version and enabled mod_dingling and compiled. Everything went through and able to establish call to an extension if I configure that extension number in "client" profile. What I am trying to do is, I want to bridge or join a call coming from GTalk to an existing conference in FreeSwitch. For this purpose I configured a different number on "client profile" and created a dial-plan for this number to 'park' the call first before trying to join to the conference. Then using eventSockets I am trying to join this call to conference and issued following command. (tried with "uuid_bridge" command as well) "api uuid_transfer [Unique-ID] conference:xyz@default inline" Command is successful and also I can hear a sound that someone joined in the conference, but I didn't hear any voice at either side. I couldn't see any RTP flow as well (checked wireshark traces at FS). After sometime like 30 seconds call at GTalk is disconnected automatically. I am not sure why nothing is heard at both sides and why call got disconnected. Also tried answering the call first (after Park) and then bridging to conference, still got the same issue. Could someone please let me know if I am missing anything or need to configure in a different way for conferencing. Thanks & Regards Suneel Client.xml <profile type="client"> <param name="name" value="gmail.com"/> <param name="login" value="user(a)gmail.com/gtalk"/> <param name="password" value="password"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="123456"/> <param name="vad" value="both"/> </profile> Dial-plan.. <include> <extension name="GTalk_DP"> <condition field="caller_id_number" expression="^([^@]+)" break="never"> <action application="set" data="effective_caller_id_number=$1" /> </condition> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="proxy_media=true"/> <action application="set" data="bypass_media=true"/> <action application="park" /> </condition> </extension> </include>

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Speex Codec Settings
by Stephen Dame 17 Nov '11

17 Nov '11

I'm working on a radio station integration project, using speex 16000 into a 16000 conference with x-lite/bria and flash/red5 app. Trying to figure out how the to change the speex codec, and change Complexity.,Quality settings,VBR, CBR etc. I want to run high/quality very small <5 participant, talk show conferences. I have a client app sending qualityencode=10 to conference, but assume its converted to PCM, mixed with other streams and sent back out to other participants at the freeswitch default settings. I looked into speex source code and see some values Complexity 5, Quality 5, VBRQuality 4, AGClevel 8000 Question: Can I somehow pass these values into the call to setup the highest and force CBR, or do I need to change in codec source and recompile? It a dedicated server used for just this application. Thanks in advance Stephen

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Issue with mod_diangling during compilation
by Papineni, Suneel 17 Nov '11

17 Nov '11

Hi, I am trying to understand mod_diangling and followed documentation at http://wiki.freeswitch.org/wiki/Dingaling As I am running on Windows, Installed GnuTls and generated libgnutls-26.lib file. Added additional include directory and dependencies to iksemel library project. Now when I compile, it should generate respective DLLs right. Unfortunately I am not finding the TLS DLLs' as mentioned in the link. I tried to compile whole solution and also compiled iksemel project and mod_diangling project separately. Still I didn't find the TLS DLLs in output (Release) folder. Can someone please let me know what I am missing here. Also please let me know if I need to follow any other procedure. Note: I am using latest version of FreeSwitch from GIT and using Visual Studio 2010. Thanks & Regards Suneel

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FreeSWITCH Conference Call Today
by Michael Collins 17 Nov '11

17 Nov '11

Hello FreeSWITCHers! Here's today's agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_09 We have a few miscellaneous items to discuss and I have some updates. Also, I'm hoping to continue the discussion on SIP 101 that we started last week. I think it would be a good time to go over the Contact: header and what it's for. That will set the ground work for what the NDLB-connectile-dysfunction parameter does. Talk to you soon, -Michael

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listing conference participants
by David Wafula 17 Nov '11

17 Nov '11

Hello all, How does one get a list of participants currently in a conference? I want to do something similar like what is on http://conference.freeswitch.org regards, -- David Wafula

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