August 2011 Archives by author

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Starting: Mon Aug 1 00:13:55 MSD 2011
Ending: Wed Aug 31 23:38:51 MSD 2011
Messages: 913

• [Freeswitch-users] Open Source Billing for FreeSWITCH to get public soon ..   mashudi72 -
• [Freeswitch-users] Error SRTP unprotect   Aakash
• [Freeswitch-users] Dialed number identification service   Aakash
• [Freeswitch-users] Dialed number identification service   Aakash
• [Freeswitch-users] Dialed number identification service   Aakash
• [Freeswitch-users] Custom SIP Profile   Nazim Aghabayov
• [Freeswitch-users] SIP proxy collect DTMF using FS   Gohar Ahmed
• [Freeswitch-users] Look n8nrcuv   Max Alex
• [Freeswitch-users] Problem receiving fax   Antonio
• [Freeswitch-users] Problem receiving fax   Antonio
• [Freeswitch-users] Problem receiving fax   Antonio
• [Freeswitch-users] mod_rtmp question   Steven Ayre
• [Freeswitch-users] Help setting up SIP reg   Steven Ayre
• [Freeswitch-users] Calls not established : See the error details   Steven Ayre
• [Freeswitch-users] Hangup hooks on B legs   Steven Ayre
• [Freeswitch-users] outboundsocket mode, bridge and bind_digit_action doesn't work   Steven Ayre
• [Freeswitch-users] Does FreeSWITCH supports Googles VP8 (WebM) Codec   Steven Ayre
• [Freeswitch-users] SIP auth challenge error   Steven Ayre
• [Freeswitch-users] outboundsocket mode, bridge and bind_digit_action doesn't work   Steven Ayre
• [Freeswitch-users] outboundsocket mode, bridge and bind_digit_action doesn't work   Steven Ayre
• [Freeswitch-users] Best way to route Large block of DIDs   Steven Ayre
• [Freeswitch-users] event   Steven Ayre
• [Freeswitch-users] event   Steven Ayre
• [Freeswitch-users] please help! Invalid file format [wav]   Steven Ayre
• [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR   Steven Ayre
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Steven Ayre
• [Freeswitch-users] error loading mod_xml_cdr   Steven Ayre
• [Freeswitch-users] May FS module Application NOT return data like FS module API?   Steven Ayre
• [Freeswitch-users] Secure connection   Steven Ayre
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Gerardo Barajas
• [Freeswitch-users] FreeSWITCH™ in Mexico   Gerardo Barajas
• [Freeswitch-users] Secure connection   Jean-Yves F. Barbier
• [Freeswitch-users] Secure connection   Jean-Yves F. Barbier
• [Freeswitch-users] Help with dual IP gateways   Abdul Basit
• [Freeswitch-users] call_timeout   Federico Beffa
• [Freeswitch-users] call_timeout   Federico Beffa
• [Freeswitch-users] call_timeout   Federico Beffa
• [Freeswitch-users] call_timeout   Federico Beffa
• [Freeswitch-users] call_timeout   Federico Beffa
• [Freeswitch-users] Problem receiving fax   Christian Benke
• [Freeswitch-users] Fetching user-variables   Christian Benke
• [Freeswitch-users] Fetching user-variables   Christian Benke
• [Freeswitch-users] Billing solutions   Cesar Bermudez
• [Freeswitch-users] Open Source Billing for FreeSWITCH to get public soon ..   Muhammad Naseer Bhatti
• [Freeswitch-users] FS Status Command Description   Leonardo P. Bidinoto
• [Freeswitch-users] Mod event socket error   Leonardo P. Bidinoto
• [Freeswitch-users] Meetup Paris?   Michal Bielicki
• [Freeswitch-users] Meetup Paris?   Michal Bielicki
• [Freeswitch-users] Building Freeswitch + FreeTDM + LibSng-ISDN RPMs   Michal Bielicki
• [Freeswitch-users] strdup error on solaris   Michal Bielicki
• [Freeswitch-users] Making Freeswitch listen on 2 IP addresses   Michal Bielicki
• [Freeswitch-users] AMR transcoding   Michal Bielicki
• [Freeswitch-users] Variables in Dialplan - Problem with getting Variables from User Directory   Michal Bielicki
• [Freeswitch-users] media proxy   Jay Binks
• [Freeswitch-users] H264 packetization with echo app   Dan-Cristian Bogos
• [Freeswitch-users] H264 packetization with echo app   Dan-Cristian Bogos
• [Freeswitch-users] DTMF bleed through in conference   David Bolen
• [Freeswitch-users] DTMF bleed through in conference   David Bolen
• [Freeswitch-users] DTMF bleed through in conference   David Bolen
• [Freeswitch-users] DTMF bleed through in conference   David Bolen
• [Freeswitch-users] Delete voicemail greeting?   Yehavi Bourvine
• [Freeswitch-users] 3   Dave Bracken
• [Freeswitch-users] Newbie - error when calling with new Yealink T20P phone   Charles Bujold
• [Freeswitch-users] Newbie - external originating leg has no sound when calling   Charles Bujold
• [Freeswitch-users] Newbie - External Phone setup question   Charles Bujold
• [Freeswitch-users] Strange problem with Freeswitch and external extension   Charles Bujold
• [Freeswitch-users] Problem with freeswith and a Digium card   Caez
• [Freeswitch-users] Freeswitch Hangup Event Tracking   Roger Castaldo
• [Freeswitch-users] Freeswitch Hangup Event Tracking   Roger Castaldo
• [Freeswitch-users] having problems building mono 2.10   Roger Castaldo
• [Freeswitch-users] FS for SIP registrar server   Dome Charoenyost
• [Freeswitch-users] FS for SIP registrar server   Dome Charoenyost
• [Freeswitch-users] g729 licenses usage   Julien Chavanton
• [Freeswitch-users] g.729 licence usage   Julien Chavanton
• [Freeswitch-users] freeswitch.org is down   Yungwei Chen
• [Freeswitch-users] Making changes to dialplans and users using API commands   Yungwei Chen
• [Freeswitch-users] voicemail is not saved   Yungwei Chen
• [Freeswitch-users] voicemail doesn't work   Yungwei Chen
• [Freeswitch-users] voicemail doesn't work   Yungwei Chen
• [Freeswitch-users] How to save voicemail msgs to a db   Yungwei Chen
• [Freeswitch-users] How to save voicemail msgs to a db   Yungwei Chen
• [Freeswitch-users] Secure connection   Chrisbware
• [Freeswitch-users] Help setting up SIP reg   Michael Collins
• [Freeswitch-users] Help setting up SIP reg   Michael Collins
• [Freeswitch-users] Help setting up SIP reg   Michael Collins
• [Freeswitch-users] Limit number of voicemails   Michael Collins
• [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to dialplan extebsion   Michael Collins
• [Freeswitch-users] (no subject)   Michael Collins
• [Freeswitch-users] (no subject)   Michael Collins
• [Freeswitch-users] (no subject)   Michael Collins
• [Freeswitch-users] (no subject)   Michael Collins
• [Freeswitch-users] (no subject)   Michael Collins
• [Freeswitch-users] (no subject)   Michael Collins
• [Freeswitch-users] conference caller id number&name   Michael Collins
• [Freeswitch-users] Calls not established : See the error details   Michael Collins
• [Freeswitch-users] Help setting up SIP reg   Michael Collins
• [Freeswitch-users] Freeswitch Hangup Event Tracking   Michael Collins
• [Freeswitch-users] Calls not established : See the error details   Michael Collins
• [Freeswitch-users] Hangup hooks on B legs   Michael Collins
• [Freeswitch-users] Help setting up SIP reg   Michael Collins
• [Freeswitch-users] FreeSWITCH Conference Call Today   Michael Collins
• [Freeswitch-users] Help setting up SIP reg   Michael Collins
• [Freeswitch-users] Hangup hooks on B legs   Michael Collins
• [Freeswitch-users] Hangup hooks on B legs   Michael Collins
• [Freeswitch-users] Last Minute ClueCon Items: Hotel, Parking, Directions, Speakers   Michael Collins
• [Freeswitch-users] api_hangup_hook and hupall   Michael Collins
• [Freeswitch-users] Hangup hooks on B legs   Michael Collins
• [Freeswitch-users] Hangup hooks on B legs   Michael Collins
• [Freeswitch-users] Hangup hooks on B legs   Michael Collins
• [Freeswitch-users] outboundsocket mode, bridge and bind_digit_action doesn't work   Michael Collins
• [Freeswitch-users] outboundsocket mode, bridge and bind_digit_action doesn't work   Michael Collins
• [Freeswitch-users] Problem with XML CDR   Michael Collins
• [Freeswitch-users] outgoing SIP profiles setup   Michael Collins
• [Freeswitch-users] conference count   Michael Collins
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Michael Collins
• [Freeswitch-users] Playback .wma/.wmv   Michael Collins
• [Freeswitch-users] Running Post-Bridge scripts   Michael Collins
• [Freeswitch-users] DTMF issue when using execute_extension with play_and_get_digits   Michael Collins
• [Freeswitch-users] Conference calls through web-interface with moderation using FreeSWITCH?   Michael Collins
• [Freeswitch-users] please help!!! playing tone only when 3rd party joins in bridging conference   Michael Collins
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Michael Collins
• [Freeswitch-users] DTMF bleed through in conference   Michael Collins
• [Freeswitch-users] ORIGINATE_DISPOSITION   Michael Collins
• [Freeswitch-users] ORIGINATE_DISPOSITION   Michael Collins
• [Freeswitch-users] ORIGINATE_DISPOSITION   Michael Collins
• [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua   Michael Collins
• [Freeswitch-users] ORIGINATE_DISPOSITION   Michael Collins
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Michael Collins
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Michael Collins
• [Freeswitch-users] FreeSWITCH Conference Call Today   Michael Collins
• [Freeswitch-users] Paging with Polycom Phones   Michael Collins
• [Freeswitch-users] Inband DTMF when rfc2833 negotiated?   Michael Collins
• [Freeswitch-users] Problem with receiving a NOTIFY after sending a SUBSCRIBE request   Michael Collins
• [Freeswitch-users] g729 licenses usage   Michael Collins
• [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua   Michael Collins
• [Freeswitch-users] Paging with Polycom Phones   Michael Collins
• [Freeswitch-users] mod_spidermonkey loading error   Michael Collins
• [Freeswitch-users] voicemail doesn't work   Michael Collins
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Michael Collins
• [Freeswitch-users] Bad port when using rport   Michael Collins
• [Freeswitch-users] Simulating Asterisk dial(x:y:z)   Michael Collins
• [Freeswitch-users] Simulating Asterisk dial(x:y:z)   Michael Collins
• [Freeswitch-users] Dynamically switching DTMF mode?   Michael Collins
• [Freeswitch-users] mod_spidermonkey loading error   Michael Collins
• [Freeswitch-users] Great ClueCon!   Michael Collins
• [Freeswitch-users] Best tabletop conference phone   Michael Collins
• [Freeswitch-users] VegaStream Contacts   Michael Collins
• [Freeswitch-users] INFO: Wiki update   Michael Collins
• [Freeswitch-users] INFO: Wiki update   Michael Collins
• [Freeswitch-users] mod_spidermonkey loading error   Michael Collins
• [Freeswitch-users] Fetching user-variables   Michael Collins
• [Freeswitch-users] enterprise deployment not working   Michael Collins
• [Freeswitch-users] enterprise deployment not working   Michael Collins
• [Freeswitch-users] mod_spidermonkey loading error   Michael Collins
• [Freeswitch-users] Newbie - external originating leg has no sound when calling   Michael Collins
• [Freeswitch-users] FreeSWITCH Janitorial Items   Michael Collins
• [Freeswitch-users] ClueCon 2011 - Assistance w/ posting videos   Michael Collins
• [Freeswitch-users] intercept sethanguphook failure   Michael Collins
• [Freeswitch-users] originate through dialplan?   Michael Collins
• [Freeswitch-users] regex help   Michael Collins
• [Freeswitch-users] freeswitch php   Michael Collins
• [Freeswitch-users] Freeswitch say digits too quick   Michael Collins
• [Freeswitch-users] Freeswitch say digits too quick   Michael Collins
• [Freeswitch-users] freeswitch php   Michael Collins
• [Freeswitch-users] FreeSWITCH Conference Call Today   Michael Collins
• [Freeswitch-users] Newbie - External Phone setup question   Michael Collins
• [Freeswitch-users] call_timeout   Michael Collins
• [Freeswitch-users] Parsing Error on conference_set_auto_outcall   Michael Collins
• [Freeswitch-users] PRI to VoIP gateway devices   Michael Collins
• [Freeswitch-users] Start Conference with first user   Michael Collins
• [Freeswitch-users] Inband DTMF and bind_meta_app problem again   Michael Collins
• [Freeswitch-users] Dialed number identification service   Michael Collins
• [Freeswitch-users] Session ends unexpectedly during record dial plan usage   Michael Collins
• [Freeswitch-users] Strange problem with Freeswitch and external extension   Michael Collins
• [Freeswitch-users] Multiple Tone Detect   Michael Collins
• [Freeswitch-users] freeswitch php   Michael Collins
• [Freeswitch-users] Session ends unexpectedly during record dial plan usage   Michael Collins
• [Freeswitch-users] FreeSWITCH Conference Call Today   Michael Collins
• [Freeswitch-users] FS behind an IPCOP firewall   Michael Collins
• [Freeswitch-users] DTMF from Nortel BCM 400 to Freeswitch   Michael Collins
• [Freeswitch-users] Error compling jsxml.c   Jerre Cope
• [Freeswitch-users] Notify message to Gateway   Marc de Corny
• [Freeswitch-users] Hold and BLF - disable flashing or pickup for held call?   Alex Crow
• [Freeswitch-users] Problem with freeswith and a Digium card   Alex Crow
• [Freeswitch-users] Problem with freeswith and a Digium card   Alex Crow
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   Tihomir Culjaga
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   Tihomir Culjaga
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   Tihomir Culjaga
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   Tihomir Culjaga
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   Tihomir Culjaga
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   Tihomir Culjaga
• [Freeswitch-users] FS performance using ESL   Tihomir Culjaga
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   Tihomir Culjaga
• [Freeswitch-users] FS terminated maybe due to memory leak?   Tihomir Culjaga
• [Freeswitch-users] FS performance using ESL   Tihomir Culjaga
• [Freeswitch-users] FS performance using ESL   Tihomir Culjaga
• [Freeswitch-users] FS performance using ESL   Tihomir Culjaga
• [Freeswitch-users] FS performance using ESL   Tihomir Culjaga
• [Freeswitch-users] Help with choppy audio after attended transfer   Chris Cureau
• [Freeswitch-users] Help with choppy audio after attended transfer   Chris Cureau
• [Freeswitch-users] Commercial support   Chris Cureau
• [Freeswitch-users] Cancel sent on outgoing call completion   Chris Cureau
• [Freeswitch-users] Sofia not taking new ipaddress from local ip '127.0.0.1'   Michel Daggelinckx
• [Freeswitch-users] Meetup Paris?   Michel Daggelinckx
• [Freeswitch-users] How to for video call ?   Nandy Dagondon
• [Freeswitch-users] Fidelio   Nandy Dagondon
• [Freeswitch-users] Fidelio   Nandy Dagondon
• [Freeswitch-users] How to for video call ?   Nandy Dagondon
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Nandy Dagondon
• [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux   Nandy Dagondon
• [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux   Nandy Dagondon
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Nandy Dagondon
• [Freeswitch-users] Help with dual IP gateways   Nandy Dagondon
• [Freeswitch-users] Help with dual IP gateways   Nandy Dagondon
• [Freeswitch-users] DTMF issue when using execute_extension with play_and_get_digits   Nandy Dagondon
• [Freeswitch-users] voicemail is not saved   Nandy Dagondon
• [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip   Nandy Dagondon
• [Freeswitch-users] Playback .wma/.wmv   Stephen Dame
• [Freeswitch-users] Limit number of voicemails   Sascha Daniels
• [Freeswitch-users] Limit number of voicemails   Sascha Daniels
• [Freeswitch-users] Limit number of voicemails   Sascha Daniels
• [Freeswitch-users] vm-alternate-greet-id is ignored   Sascha Daniels
• [Freeswitch-users] CallWithUs   Darcy
• [Freeswitch-users] How to get devices registered with skinny protocol   Manjiri Deshpande
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Nestor A Diaz
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Nestor A Diaz
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Nestor A Diaz
• [Freeswitch-users] Sip Headers advice (Not parsing properly)   Dissident
• [Freeswitch-users] Calls not established : See the error details   Dissident
• [Freeswitch-users] Goip GSM Gateway works great with FreeSwitch!   Dissident
• [Freeswitch-users] Goip GSM Gateway works great with FreeSwitch!   Dissident
• [Freeswitch-users] Condition based on custom sip header   Dissident
• [Freeswitch-users] Question on caller id   Seven Du
• [Freeswitch-users] Question on caller id   Seven Du
• [Freeswitch-users] Does FreeSWITCH supports Googles VP8 (WebM) Codec   Seven Du
• [Freeswitch-users] openssl deprecated on OS X Lion and MacPorts with latest Git head?   Seven Du
• [Freeswitch-users] please help! Invalid file format [wav]   Seven Du
• [Freeswitch-users] strdup error on solaris   Seven Du
• [Freeswitch-users] strdup error on solaris   Seven Du
• [Freeswitch-users] Variables in Dialplan - Problem with getting Variables from User Directory   Silvio Escher
• [Freeswitch-users] Variables in Dialplan - Problem with getting Variables from User Directory   Silvio Escher
• [Freeswitch-users] Problem with receiving a NOTIFY after sending a SUBSCRIBE request   Tom Fayette
• [Freeswitch-users] Error Compiling esl - php   Joe Flemmings
• [Freeswitch-users] Error Compiling esl - php   Joe Flemmings
• [Freeswitch-users] mod_rtmp & flex client   Matthew Fong
• [Freeswitch-users] mod_rtmp & flex client   Matthew Fong
• [Freeswitch-users] mod_rtmp & flex client   Matthew Fong
• [Freeswitch-users] Grandstreams and TLS   Joshua Foshee
• [Freeswitch-users] Error compling jsxml.c   D Foulkes
• [Freeswitch-users] Help with dual IP gateways   Mario G
• [Freeswitch-users] Help with dual IP gateways   Mario G
• [Freeswitch-users] Problem with freeswith and a Digium card   Sébastien Gay
• [Freeswitch-users] Problem with freeswith and a Digium card   Sébastien Gay
• [Freeswitch-users] TTS/unimrcp engines   Hector Geraldino
• [Freeswitch-users] TTS/unimrcp engines   Hector Geraldino
• [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed   Hector Geraldino
• [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed   Hector Geraldino
• [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed   Hector Geraldino
• [Freeswitch-users] mod_event_socket Outbound connection in Java.   Hector Geraldino
• [Freeswitch-users] How to do "search" in freeswitch-users Archives?   Hector Geraldino
• [Freeswitch-users] FS/MRCP/SIP issue....   Mike Ghicas
• [Freeswitch-users] FS/MRCP/SIP issue....   Mike Ghicas
• [Freeswitch-users] ORIGINATE_DISPOSITION   Michael Giagnocavo
• [Freeswitch-users] having problems building mono 2.10   Michael Giagnocavo
• [Freeswitch-users] having problems building mono 2.10   Michael Giagnocavo
• [Freeswitch-users] problem using mod_managed under linux   Michael Giagnocavo
• [Freeswitch-users] problem using mod_managed under linux   Michael Giagnocavo
• [Freeswitch-users] Running Post-Bridge scripts   Alec Glassford
• [Freeswitch-users] Record-Route Header Rejection   Loy Glenn
• [Freeswitch-users] FreeSWITCH™ in Mexico   Fredy Gonzales
• [Freeswitch-users] SIP proxy collect DTMF using FS   Sam Govind
• [Freeswitch-users] SIP proxy collect DTMF using FS   Sam Govind
• [Freeswitch-users] SIP proxy collect DTMF using FS   Sam Govind
• [Freeswitch-users] ESL Inbound - what would be the best known way to wait in XML dialplan for any command execution via ESL?   Gabriel Gunderson
• [Freeswitch-users] Error compling jsxml.c   Gabriel Gunderson
• [Freeswitch-users] Mod_Managed Mono is not following the system LD Path   R H
• [Freeswitch-users] Mod_Managed Mono is not following the system LD Path   R H
• [Freeswitch-users] Core dump - When using user_recurse_variables in bridge   Nagalenoj H.
• [Freeswitch-users] DTMF issue when using execute_extension with play_and_get_digits   Nagalenoj H.
• [Freeswitch-users] DTMF issue when using execute_extension with play_and_get_digits   Nagalenoj H.
• [Freeswitch-users] Freetdm bridge to SIP DTMF problem   Robert Hadley
• [Freeswitch-users] How to save voicemail msgs to a db   Robert Hadley
• [Freeswitch-users] How to save voicemail msgs to a db   Robert Hadley
• [Freeswitch-users] Still having problems making calls to phones behind double nat   Gavin Henry
• [Freeswitch-users] openssl deprecated on OS X Lion and MacPorts with latest Git head?   Gavin Henry
• [Freeswitch-users] call_timeout   Gavin Henry
• [Freeswitch-users] rtp natting   Gavin Henry
• [Freeswitch-users] Limit number of voicemails   Robert Huddleston
• [Freeswitch-users] Limit number of voicemails   Robert Huddleston
• [Freeswitch-users] Fidelio   Robert Huddleston
• [Freeswitch-users] SIP auth challenge error   Robert Huddleston
• [Freeswitch-users] SIP auth challenge error   Robert Huddleston
• [Freeswitch-users] CallWithUs   Robert Huddleston
• [Freeswitch-users] CallWithUs   Robert Huddleston
• [Freeswitch-users] Setup with Bandwidth.com   Robert Huddleston
• [Freeswitch-users] 400 Bad Record-Route Header   Matt Hunter
• [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux   Rob Hutton
• [Freeswitch-users] mod_dingaling delay buildup   Rob Hutton
• [Freeswitch-users] Slightly OT - Freeswitch and Vitelity?   Rob Hutton
• [Freeswitch-users] Multiple Tone Detect   Rob Hutton
• [Freeswitch-users] Looking up variable from config   Rob Hutton
• [Freeswitch-users] Looking up variable from config   Rob Hutton
• [Freeswitch-users] Dialed number identification service   Rob Hutton
• [Freeswitch-users] Dialed number identification service   Rob Hutton
• [Freeswitch-users] Problem with FIFO music and chime   Tomasz Hyziak
• [Freeswitch-users] Problem with FIFO music and chime   Tomasz Hyziak
• [Freeswitch-users] Cannot transfer incoming call   Tomasz Hyziak
• [Freeswitch-users] originate through dialplan?   Royce Mitchell III
• [Freeswitch-users] originate through dialplan?   Royce Mitchell III
• [Freeswitch-users] Multiple Registration   DJB International
• [Freeswitch-users] Multiple Registration   DJB International
• [Freeswitch-users] SYSTEM_SHUTDOWN   DJB International
• [Freeswitch-users] Problem with receiving a NOTIFY after sending a SUBSCRIBE request   Lorem Ipsum
• [Freeswitch-users] Fidelio   Hugh Irvine
• [Freeswitch-users] How to authencticate with mod_rtmp?   Jack
• [Freeswitch-users] How to authencticate with mod_rtmp?   Jack
• [Freeswitch-users] SIP auth challenge error   John
• [Freeswitch-users] mod_dingaling delay buildup   Jon
• [Freeswitch-users] mod_dingaling delay buildup   Jon
• [Freeswitch-users] Hangup hooks on B legs   Isaac Jurado
• [Freeswitch-users] Hangup hooks on B legs   Isaac Jurado
• [Freeswitch-users] Hangup hooks on B legs   Isaac Jurado
• [Freeswitch-users] CDR ordering?   Isaac Jurado
• [Freeswitch-users] Hangup hooks on B legs   Isaac Jurado
• [Freeswitch-users] Hangup hooks on B legs   Isaac Jurado
• [Freeswitch-users] Hangup hooks on B legs   Isaac Jurado
• [Freeswitch-users] Hangup hooks on B legs   Isaac Jurado
• [Freeswitch-users] Hangup hooks on B legs   Isaac Jurado
• [Freeswitch-users] Hangup hooks on B legs   Isaac Jurado
• [Freeswitch-users] Hangup hooks on B legs   Isaac Jurado
• [Freeswitch-users] Delete voicemail greeting?   Isaac Jurado
• [Freeswitch-users] Delete voicemail greeting?   Isaac Jurado
• [Freeswitch-users] How to start IVR script via dialplan by making an outbound call   Isaac Jurado
• [Freeswitch-users] How to save voicemail msgs to a db   Isaac Jurado
• [Freeswitch-users] How to save voicemail msgs to a db   Isaac Jurado
• [Freeswitch-users] How to save voicemail msgs to a db   Isaac Jurado
• [Freeswitch-users] My new web utility for FreeSWITCH   Valery Kalinin
• [Freeswitch-users] playing value with mod_say & getting DTMF input at the same time   Mi Ke
• [Freeswitch-users] playing value with mod_say & getting DTMF input at the same time   Mi Ke
• [Freeswitch-users] Fw: Re: playing value with mod_say & getting DTMF input at the same time   Mi Ke
• [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua   Mi Ke
• [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua   Mi Ke
• [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua   Mi Ke
• [Freeswitch-users] Session ends unexpectedly during record dial plan usage   Adam Kelloway
• [Freeswitch-users] Session ends unexpectedly during record dial plan usage   Adam Kelloway
• [Freeswitch-users] Making Freeswitch listen on 2 IP addresses   Pete Kelly
• [Freeswitch-users] Making Freeswitch listen on 2 IP addresses   Pete Kelly
• [Freeswitch-users] mod_nibblebill mp3   Pete Kelly
• [Freeswitch-users] Making Freeswitch listen on 2 IP addresses   Pete Kelly
• [Freeswitch-users] Setup with Bandwidth.com   Travis Kemen
• [Freeswitch-users] mod_conference - 2-digit DTMF and events   Michael Kemp
• [Freeswitch-users] mod_conference - 2-digit DTMF and events   Michael Kemp
• [Freeswitch-users] Custom SIP Profile   HM Kias
• [Freeswitch-users] freeswitch.org is down   HM Kias
• [Freeswitch-users] DTMF bleed through in conference   Kristian Kielhofner
• [Freeswitch-users] rtp natting   Kristian Kielhofner
• [Freeswitch-users] having problems building mono 2.10   Stefan Knoblich
• [Freeswitch-users] DNS SRV records   Micha Knop
• [Freeswitch-users] AMR transcoding   Michal Kopacki
• [Freeswitch-users] AMR transcoding   Michal Kopacki
• [Freeswitch-users] AMR transcoding   Michal Kopacki
• [Freeswitch-users] AMR transcoding   Michal Kopacki
• [Freeswitch-users] Newbie - External Phone setup question   Michal Kopacki
• [Freeswitch-users] DTMF when speaking   Boris Kovalenko
• [Freeswitch-users] DTMF when speaking   Boris Kovalenko
• [Freeswitch-users] freeswitch.org is down   Kris
• [Freeswitch-users] [Freeswitch-dev] Meetup Paris?   Prashant Lamba
• [Freeswitch-users] Question -- DTMF from RFC2833 to SIP INFO   Dan Lan
• [Freeswitch-users] Connecting Call Between Two SIP Trunks   Dan Lan
• [Freeswitch-users] Connecting Call Between Two SIP Trunks   Dan Lan
• [Freeswitch-users] Disable Allow Events in Fs   Simon Leck
• [Freeswitch-users] media proxy   Simon Leck
• [Freeswitch-users] Freeswitch ignoring bye   Simon Leck
• [Freeswitch-users] Problem with freeswith and a Digium card   François Legal
• [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip   Bryan Lemon
• [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip   Bryan Lemon
• [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip   Bryan Lemon
• [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip   Bryan Lemon
• [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip   Bryan Lemon
• [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip   Bryan Lemon
• [Freeswitch-users] (no subject)   Jeff Lenk
• [Freeswitch-users] FS Status Command Description   Jeff Lenk
• [Freeswitch-users] freeswitch.org is down   Jeff Lenk
• [Freeswitch-users] FS terminated maybe due to memory leak?   Jeff Lenk
• [Freeswitch-users] Playback .wma/.wmv   Jeff Lenk
• [Freeswitch-users] incompatible destination because of crypto incoming calls   Jeff Lenk
• [Freeswitch-users] Weird: Sofia-sip failed to be compiled on Visual Studio 2008 but it was OK three weeks ago   Jeff Lenk
• [Freeswitch-users] call_timeout   Jeff Lenk
• [Freeswitch-users] call_timeout   Jeff Lenk
• [Freeswitch-users] Parsing Error on conference_set_auto_outcall   Jeff Lenk
• [Freeswitch-users] How to save voicemail msgs to a db   Chad Phillips -- Apartment Lines
• [Freeswitch-users] Invitation to connect on LinkedIn   Martin Armstrong via LinkedIn
• [Freeswitch-users] Invitation to connect on LinkedIn   Martin Armstrong via LinkedIn
• [Freeswitch-users] Commercial support   Patrick Lists
• [Freeswitch-users] 400 Bad Record-Route Header   Patrick Lists
• [Freeswitch-users] Secure connection   Patrick Lists
• [Freeswitch-users] Best tabletop conference phone   Patrick Lists
• [Freeswitch-users] mod_event_socket Outbound connection in Java.   Suresh M
• [Freeswitch-users] mod_event_socket Outbound connection in Java.   Suresh M
• [Freeswitch-users] How to use ESL event   David Ma
• [Freeswitch-users] Receiving junk ESL events   David Ma
• [Freeswitch-users] mod_rtmp question   Madovsky
• [Freeswitch-users] mod_rtmp question   Madovsky
• [Freeswitch-users] ringing status and mod_rtmp   Madovsky
• [Freeswitch-users] about fax status   Madovsky
• [Freeswitch-users] Calling FMS with mod_rtmp   Madovsky
• [Freeswitch-users] ringing status and mod_rtmp   Madovsky
• [Freeswitch-users] Hangup hooks on B legs   Avi Marcus
• [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to dialplan extebsion   Avi Marcus
• [Freeswitch-users] Hangup hooks on B legs   Avi Marcus
• [Freeswitch-users] Hangup hooks on B legs   Avi Marcus
• [Freeswitch-users] CDR ordering?   Avi Marcus
• [Freeswitch-users] Problem with XML CDR   Avi Marcus
• [Freeswitch-users] FS for SIP registrar server   Avi Marcus
• [Freeswitch-users] Best way to route Large block of DIDs   Avi Marcus
• [Freeswitch-users] playing value with mod_say & getting DTMF input at the same time   Avi Marcus
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Avi Marcus
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Avi Marcus
• [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux   Avi Marcus
• [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux   Avi Marcus
• [Freeswitch-users] rewind and fast forward while playing a file   Avi Marcus
• [Freeswitch-users] odbc basic_calls   Avi Marcus
• [Freeswitch-users] odbc basic_calls   Avi Marcus
• [Freeswitch-users] Making changes to dialplans and users using API commands   Avi Marcus
• [Freeswitch-users] Running Post-Bridge scripts   Avi Marcus
• [Freeswitch-users] collect dtmf and store in channel variable   Avi Marcus
• [Freeswitch-users] collect dtmf and store in channel variable   Avi Marcus
• [Freeswitch-users] collect dtmf and store in channel variable   Avi Marcus
• [Freeswitch-users] collect dtmf and store in channel variable   Avi Marcus
• [Freeswitch-users] odbc basic_calls   Avi Marcus
• [Freeswitch-users] odbc basic_calls   Avi Marcus
• [Freeswitch-users] odbc basic_calls   Avi Marcus
• [Freeswitch-users] odbc basic_calls   Avi Marcus
• [Freeswitch-users] odbc basic_calls   Avi Marcus
• [Freeswitch-users] odbc basic_calls   Avi Marcus
• [Freeswitch-users] Secure connection   Avi Marcus
• [Freeswitch-users] Inband DTMF when rfc2833 negotiated?   Avi Marcus
• [Freeswitch-users] Goip GSM Gateway works great with FreeSwitch!   Avi Marcus
• [Freeswitch-users] Billing solutions   Avi Marcus
• [Freeswitch-users] DTMF bleed through in conference   Avi Marcus
• [Freeswitch-users] Multiple Registration   Avi Marcus
• [Freeswitch-users] Inband DTMF when rfc2833 negotiated?   Avi Marcus
• [Freeswitch-users] Freeswitch ignoring bye   Avi Marcus
• [Freeswitch-users] Goip GSM Gateway works great with FreeSwitch!   Avi Marcus
• [Freeswitch-users] Ignore INFO DTMF   Avi Marcus
• [Freeswitch-users] nf_ct_sip dropped packets?   Avi Marcus
• [Freeswitch-users] Making Freeswitch listen on 2 IP addresses   Avi Marcus
• [Freeswitch-users] How to save voicemail msgs to a db   Avi Marcus
• [Freeswitch-users] FREE webinar video about Auto-Dialer Business Model (Telemarketing)   Kolmisoft Marketing
• [Freeswitch-users] rtp natting   Chris Martineau
• [Freeswitch-users] rtp natting   Chris Martineau
• [Freeswitch-users] How to for video call ?   Giovanni Maruzzelli
• [Freeswitch-users] freetdm error message   Giovanni Maruzzelli
• [Freeswitch-users] Meetup Paris?   Giovanni Maruzzelli
• [Freeswitch-users] any IVR example in C/C++?   Giovanni Maruzzelli
• [Freeswitch-users] any IVR example in C/C++?   Giovanni Maruzzelli
• [Freeswitch-users] [Newbie] Unable to play .mp3 file   Giovanni Maruzzelli
• [Freeswitch-users] [Freeswitch-dev] Meetup Paris?   Giovanni Maruzzelli
• [Freeswitch-users] Forcing CN offer   Alex Massover
• [Freeswitch-users] Forcing CN offer   Alex Massover
• [Freeswitch-users] Forcing CN offer   Alex Massover
• [Freeswitch-users] Condition based on custom sip header   Alex Massover
• [Freeswitch-users] Condition based on custom sip header   Alex Massover
• [Freeswitch-users] Condition based on custom sip header   Alex Massover
• [Freeswitch-users] incompatible destination because of crypto on incoming calls   Anestis Mavro
• [Freeswitch-users] incompatible destination because of crypto incoming calls   Anestis Mavro
• [Freeswitch-users] Radius/Authentication/Authorization   Mike McCauley
• [Freeswitch-users] problem using mod_managed under linux   Tim McQueen
• [Freeswitch-users] FreeSwitch Utilizes 100% of CPU Sometimes When User Quits Conference   Erwin Merioles
• [Freeswitch-users] FreeSwitch Utilizes 100% of CPU Sometimes When User Quits Conference   Erwin Merioles
• [Freeswitch-users] Fidelio   João Mesquita
• [Freeswitch-users] Fidelio   João Mesquita
• [Freeswitch-users] Fidelio   João Mesquita
• [Freeswitch-users] Does FreeSWITCH supports Googles VP8 (WebM) Codec   João Mesquita
• [Freeswitch-users] FreeSWITCH™ in Mexico   João Mesquita
• [Freeswitch-users] How to save voicemail msgs to a db   João Mesquita
• [Freeswitch-users] FreeSWITCH™ in Mexico   João Mesquita
• [Freeswitch-users] Best tabletop conference phone   john Millican
• [Freeswitch-users] How to start IVR script via dialplan by making an outbound call   Brad Mina
• [Freeswitch-users] Connecting Call Between Two SIP Trunks   Brad Mina
• [Freeswitch-users] Setup with Bandwidth.com   Brad Mina
• [Freeswitch-users] Sip Headers advice (Not parsing properly)   Anthony Minessale
• [Freeswitch-users] Help with choppy audio after attended transfer   Anthony Minessale
• [Freeswitch-users] ESL: not bein able to determine when SIP gateway is down when originating a call   Anthony Minessale
• [Freeswitch-users] Sip Headers advice (Not parsing properly)   Anthony Minessale
• [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to dialplan extebsion   Anthony Minessale
• [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to dialplan extebsion   Anthony Minessale
• [Freeswitch-users] Receiving junk ESL events   Anthony Minessale
• [Freeswitch-users] Core dump - When using user_recurse_variables in bridge   Anthony Minessale
• [Freeswitch-users] commit broke one of my trunks   Anthony Minessale
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Anthony Minessale
• [Freeswitch-users] commit broke one of my trunks   Anthony Minessale
• [Freeswitch-users] odbc basic_calls   Anthony Minessale
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Anthony Minessale
• [Freeswitch-users] FS performance using ESL   Anthony Minessale
• [Freeswitch-users] odbc basic_calls   Anthony Minessale
• [Freeswitch-users] odbc basic_calls   Anthony Minessale
• [Freeswitch-users] collect dtmf and store in channel variable   Anthony Minessale
• [Freeswitch-users] odbc basic_calls   Anthony Minessale
• [Freeswitch-users] odbc basic_calls   Anthony Minessale
• [Freeswitch-users] FS performance using ESL   Anthony Minessale
• [Freeswitch-users] FS performance using ESL   Anthony Minessale
• [Freeswitch-users] Mod event socket error   Anthony Minessale
• [Freeswitch-users] odbc basic_calls   Anthony Minessale
• [Freeswitch-users] ORIGINATE_DISPOSITION   Anthony Minessale
• [Freeswitch-users] FS performance using ESL   Anthony Minessale
• [Freeswitch-users] ORIGINATE_DISPOSITION   Anthony Minessale
• [Freeswitch-users] ORIGINATE_DISPOSITION   Anthony Minessale
• [Freeswitch-users] ORIGINATE_DISPOSITION   Anthony Minessale
• [Freeswitch-users] any IVR example in C/C++?   Anthony Minessale
• [Freeswitch-users] ORIGINATE_DISPOSITION   Anthony Minessale
• [Freeswitch-users] DTMF bleed through in conference   Anthony Minessale
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Anthony Minessale
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Anthony Minessale
• [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip   Anthony Minessale
• [Freeswitch-users] Problem with FIFO music and chime   Anthony Minessale
• [Freeswitch-users] g729 licenses usage   Anthony Minessale
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Anthony Minessale
• [Freeswitch-users] strdup error on solaris   Anthony Minessale
• [Freeswitch-users] Problem with FIFO music and chime   Anthony Minessale
• [Freeswitch-users] Simulating Asterisk dial(x:y:z)   Anthony Minessale
• [Freeswitch-users] Original Caller ID/number after attended transfer   Anthony Minessale
• [Freeswitch-users] Original Caller ID/number after attended transfer   Anthony Minessale
• [Freeswitch-users] Freeswitch say digits too quick   Anthony Minessale
• [Freeswitch-users] How to authencticate with mod_rtmp?   Anthony Minessale
• [Freeswitch-users] FreeSwitch Utilizes 100% of CPU Sometimes When User Quits Conference   Anthony Minessale
• [Freeswitch-users] Options ping timer adjustment?   Anthony Minessale
• [Freeswitch-users] H264 packetization with echo app   Anthony Minessale
• [Freeswitch-users] Forcing CN offer   Anthony Minessale
• [Freeswitch-users] odd behavior with freeswitch current   Anthony Minessale
• [Freeswitch-users] odd behavior with freeswitch current   Anthony Minessale
• [Freeswitch-users] Multiple Tone Detect   Anthony Minessale
• [Freeswitch-users] Error compling jsxml.c   Anthony Minessale
• [Freeswitch-users] Session ends unexpectedly during record dial plan usage   Anthony Minessale
• [Freeswitch-users] Condition based on custom sip header   Anthony Minessale
• [Freeswitch-users] Condition based on custom sip header   Anthony Minessale
• [Freeswitch-users] session:recordFile(-) always creates empty wav file during outbound IVR call   Anthony Minessale
• [Freeswitch-users] TTS/unimrcp engines   Jason Moran
• [Freeswitch-users] TTS/unimrcp engines   Jason Moran
• [Freeswitch-users] how to have fs interrupt a file or tts ifdigit is pressed   Jason Moran
• [Freeswitch-users] Lua not playing wav files   Yuriy Nasida
• [Freeswitch-users] Lua not playing wav files   Yuriy Nasida
• [Freeswitch-users] Lua not playing wav files   Yuriy Nasida
• [Freeswitch-users] Lua not playing wav files   Yuriy Nasida
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Moe Navid
• [Freeswitch-users] Sangoma Media Gateway   Ben Naylor
• [Freeswitch-users] Sangoma Media Gateway   Ben Naylor
• [Freeswitch-users] Sangoma Media Gateway   Ben Naylor
• [Freeswitch-users] Sangoma Media Gateway   Ben Naylor
• [Freeswitch-users] freeswitch media_proxy and zrtp   Petr Nyklicek
• [Freeswitch-users] FreeSWITCH-users Digest, Vol 62, Issue 178   Petr Nyklicek
• [Freeswitch-users] Build problem with mod_python CentOS 5.5 Python 2.7   John O'Brien
• [Freeswitch-users] Audio Problem - one way   Erin O'Meara
• [Freeswitch-users] ESL: not bein able to determine when SIP gateway is down when originating a call   Peter Olsson
• [Freeswitch-users] odbc basic_calls   Peter Olsson
• [Freeswitch-users] Help with dual IP gateways   Peter Olsson
• [Freeswitch-users] Mod event socket error   Peter Olsson
• [Freeswitch-users] Lua not playing wav files   Peter Olsson
• [Freeswitch-users] DTMF when speaking   Peter Olsson
• [Freeswitch-users] Multiple Registration   Peter Olsson
• [Freeswitch-users] Paging with Polycom Phones   Peter Olsson
• [Freeswitch-users] Great ClueCon!   Peter Olsson
• [Freeswitch-users] Session ends unexpectedly during record dial plan usage   Peter Olsson
• [Freeswitch-users] Start FreeSWITCH as a Windows service with -nonat option   Peter Olsson
• [Freeswitch-users] Freetdm bridge to SIP DTMF problem   Jakub Ouhrabka
• [Freeswitch-users] Freetdm bridge to SIP DTMF problem   Jakub Ouhrabka
• [Freeswitch-users] Freetdm bridge to SIP DTMF problem   Jakub Ouhrabka
• [Freeswitch-users] Freetdm bridge to SIP DTMF problem   Jakub Ouhrabka
• [Freeswitch-users] Freetdm bridge to SIP DTMF problem   Jakub Ouhrabka
• [Freeswitch-users] DTMF from Nortel BCM 400 to Freeswitch   Ray Pang
• [Freeswitch-users] DTMF from Nortel BCM 400 to Freeswitch   Ray Pang
• [Freeswitch-users] DTMF from Nortel BCM 400 to Freeswitch   Ray Pang
• [Freeswitch-users] Cisco Presence   Peder
• [Freeswitch-users] freeswitch.org is down   Peder
• [Freeswitch-users] Options ping timer adjustment?   Tim St. Pierre
• [Freeswitch-users] SIP/2.0 482 Request merged with Huawei-MC820/1.0.0   Eugene Prokopiev
• [Freeswitch-users] odd behavior with freeswitch current   Kevin Raison
• [Freeswitch-users] odd behavior with freeswitch current   Kevin Raison
• [Freeswitch-users] odd behavior with freeswitch current   Kevin Raison
• [Freeswitch-users] FS behind an IPCOP firewall   Praveen Ray
• [Freeswitch-users] Calling FMS with mod_rtmp   Kevin Reeves
• [Freeswitch-users] Meetup Paris?   Mathieu Rene
• [Freeswitch-users] mod_rtmp & flex client   Mathieu Rene
• [Freeswitch-users] mod_rtmp & flex client   Mathieu Rene
• [Freeswitch-users] mod_rtmp & Flex client   Mathieu Rene
• [Freeswitch-users] How to authencticate with mod_rtmp?   Mathieu Rene
• [Freeswitch-users] ringing status and mod_rtmp   Mathieu Rene
• [Freeswitch-users] Calling FMS with mod_rtmp   Mathieu Rene
• [Freeswitch-users] ringing status and mod_rtmp   Mathieu Rene
• [Freeswitch-users] FreeSwitch Utilizes 100% of CPU Sometimes When User Quits Conference   Mathieu Rene
• [Freeswitch-users] 3   Ken Rice
• [Freeswitch-users] Sip Headers advice (Not parsing properly)   Ken Rice
• [Freeswitch-users] Infolink, ServerPronto, ColoPronto, Etc <YES ITS IS OFF TOPIC>   Ken Rice
• [Freeswitch-users] freeswitch 1.0.7 required   Ken Rice
• [Freeswitch-users] freeswitch 1.0.7 required   Ken Rice
• [Freeswitch-users] Best way to route Large block of DIDs   Ken Rice
• [Freeswitch-users] outboundsocket mode, bridge and bind_digit_action doesn't work   Michael Ricordeau
• [Freeswitch-users] outboundsocket mode, bridge and bind_digit_action doesn't work   Michael Ricordeau
• [Freeswitch-users] outboundsocket mode, bridge and bind_digit_action doesn't work   Michael Ricordeau
• [Freeswitch-users] outboundsocket mode, bridge and bind_digit_action doesn't work   Michael Ricordeau
• [Freeswitch-users] mod_spidermonkey loading error   Christopher Rienzo
• [Freeswitch-users] Setup with Bandwidth.com   Robert-iPhone
• [Freeswitch-users] Sip Headers advice (Not parsing properly)   Leon de Rooij
• [Freeswitch-users] Ignore INFO DTMF   Leon de Rooij
• [Freeswitch-users] freeswitch crashed - gdb trace   Leon de Rooij
• [Freeswitch-users] Play wmv/wma streams into calls   Duvid Rottenberg
• [Freeswitch-users] Playback .wma/.wmv   Duvid Rottenberg
• [Freeswitch-users] Playback .wma/.wmv   Duvid Rottenberg
• [Freeswitch-users] Playback .wma/.wmv   Duvid Rottenberg
• [Freeswitch-users] Paging with Polycom Phones   Duvid Rottenberg
• [Freeswitch-users] Paging with Polycom Phones   Duvid Rottenberg
• [Freeswitch-users] Paging with Polycom Phones   Duvid Rottenberg
• [Freeswitch-users] Parsing Error on conference_set_auto_outcall   Duvid Rottenberg
• [Freeswitch-users] Parsing Error on conference_set_auto_outcall   Duvid Rottenberg
• [Freeswitch-users] Parsing Error on conference_set_auto_outcall   Duvid Rottenberg
• [Freeswitch-users] Building Freeswitch + FreeTDM + LibSng-ISDN RPMs   Octavio Ruiz
• [Freeswitch-users] (no subject)   Sam
• [Freeswitch-users] (no subject)   Sam
• [Freeswitch-users] (no subject)   Sam
• [Freeswitch-users] (no subject)   Sam
• [Freeswitch-users] (no subject)   Sam
• [Freeswitch-users] (no subject)   Sam
• [Freeswitch-users] conference count   Sam
• [Freeswitch-users] Hangup time is blank   Sam
• [Freeswitch-users] conference count   Sam
• [Freeswitch-users] event   Sam
• [Freeswitch-users] event   Sam
• [Freeswitch-users] event   Sam
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Sam
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Sam
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Sam
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Sam
• [Freeswitch-users] event   Sam
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Sam
• [Freeswitch-users] ORIGINATE_DISPOSITION   Sam
• [Freeswitch-users] ORIGINATE_DISPOSITION   Sam
• [Freeswitch-users] ORIGINATE_DISPOSITION   Sam
• [Freeswitch-users] ORIGINATE_DISPOSITION   Sam
• [Freeswitch-users] ORIGINATE_DISPOSITION   Sam
• [Freeswitch-users] ORIGINATE_DISPOSITION   Sam
• [Freeswitch-users] ORIGINATE_DISPOSITION   Sam
• [Freeswitch-users] ORIGINATE_DISPOSITION   Sam
• [Freeswitch-users] ORIGINATE_DISPOSITION   Sam
• [Freeswitch-users] ORIGINATE_DISPOSITION   Sam
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Sam
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Sam
• [Freeswitch-users] Simulating Asterisk dial(x:y:z)   Sam
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Sam
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Sam
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Sam
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Sam
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Sam
• [Freeswitch-users] Asterisk to FreeSWITCH migration guide   Sam
• [Freeswitch-users] Simulating Asterisk dial(x:y:z)   Sam
• [Freeswitch-users] RFC2833 and Inband DTMF   Sam
• [Freeswitch-users] play   Sam
• [Freeswitch-users] play   Sam
• [Freeswitch-users] fs_irvd   Sam
• [Freeswitch-users] freeswitch php   Sam
• [Freeswitch-users] Freeswitch say digits too quick   Sam
• [Freeswitch-users] freeswitch php   Sam
• [Freeswitch-users] Freeswitch say digits too quick   Sam
• [Freeswitch-users] freeswitch php   Sam
• [Freeswitch-users] Newbie - error when calling with new Yealink T20P phone   Juan Antonio Ibañez Santorum
• [Freeswitch-users] OS X ppc leopard "make" error message   Peter Schrock
• [Freeswitch-users] OS X PPC Leopard error on "make"   Peter Schrock
• [Freeswitch-users] Does FreeSWITCH supports Googles VP8 (WebM) Codec   Markus Siebeneicher
• [Freeswitch-users] freetdm error message   Moises Silva
• [Freeswitch-users] any IVR example in C/C++?   Moises Silva
• [Freeswitch-users] freeswitch media_proxy and zrtp   Moises Silva
• [Freeswitch-users] Sangoma Media Gateway   Moises Silva
• [Freeswitch-users] Sangoma Media Gateway   Moises Silva
• [Freeswitch-users] AMR transcoding   Moises Silva
• [Freeswitch-users] AMR transcoding   Moises Silva
• [Freeswitch-users] AMR transcoding   Moises Silva
• [Freeswitch-users] AMR transcoding   Moises Silva
• [Freeswitch-users] mod_conference - 2-digit DTMF and events   Moises Silva
• [Freeswitch-users] freetdm api - ftdm_channel_read()   Moises Silva
• [Freeswitch-users] mod_conference - 2-digit DTMF and events   Moises Silva
• [Freeswitch-users] rfc 5168 (picture fast update ) support in freeswitch.   Narendra Sirugudi
• [Freeswitch-users] mod_rtmp & Flex client   John Smith
• [Freeswitch-users] mod_rtmp & Flex client   John Smith
• [Freeswitch-users] SIP MESSAGE delivery via Freeswitch   Soomro, Amjad
• [Freeswitch-users] Sip Headers advice (Not parsing properly)   Andy Spitzer
• [Freeswitch-users] Fidelio   Peter Steinbach
• [Freeswitch-users] any IVR example in C/C++?   Campbell Steven
• [Freeswitch-users] ORIGINATE_DISPOSITION   Dmitry Sytchev
• [Freeswitch-users] Dynamically switching DTMF mode?   Dmitry Sytchev
• [Freeswitch-users] Dynamically switching DTMF mode?   Dmitry Sytchev
• [Freeswitch-users] Inband DTMF and bind_meta_app problem again   Dmitry Sytchev
• [Freeswitch-users] No sound in intercepted call leg   Dmitry Sytchev
• [Freeswitch-users] Inband DTMF and bind_meta_app problem again   Dmitry Sytchev
• [Freeswitch-users] No sound in intercepted call leg   Dmitry Sytchev
• [Freeswitch-users] Conference calls through web-interface with moderation using FreeSWITCH?   Alec Taylor
• [Freeswitch-users] Meetup Paris?   Alec Taylor
• [Freeswitch-users] Meetup Paris?   Alec Taylor
• [Freeswitch-users] Making Freeswitch listen on 2 IP addresses   Alec Taylor
• [Freeswitch-users] Ignore INFO DTMF   Mitja Thomas
• [Freeswitch-users] [solved] Ignore INFO DTMF   Mitja Thomas
• [Freeswitch-users] Start Conference with first user   Mitja Thomas
• [Freeswitch-users] Start Conference with first user   Mitja Thomas
• [Freeswitch-users] mod_com_g729 not cleaning up channels   Spencer Thomason
• [Freeswitch-users] Bad port when using rport   Spencer Thomason
• [Freeswitch-users] Best tabletop conference phone   Spencer Thomason
• [Freeswitch-users] Fidelio   Saugort Dario Garcia Tovar
• [Freeswitch-users] Problem with freeswith and a Digium card   Saugort Dario Garcia Tovar
• [Freeswitch-users] Does FreeSWITCH supports Googles VP8 (WebM) Codec   Steve Underwood
• [Freeswitch-users] Best tabletop conference phone   Steve Underwood
• [Freeswitch-users] Best tabletop conference phone   Steve Underwood
• [Freeswitch-users] AMR transcoding   Steve Underwood
• [Freeswitch-users] AMR transcoding   Steve Underwood
• [Freeswitch-users] about fax status   Steve Underwood
• [Freeswitch-users] AMR transcoding   Steve Underwood
• [Freeswitch-users] AMR transcoding   Steve Underwood
• [Freeswitch-users] Best way to route Large block of DIDs   Luis F Urrea
• [Freeswitch-users] Best tabletop conference phone   Luis F Urrea
• [Freeswitch-users] Best tabletop conference phone   Luis F Urrea
• [Freeswitch-users] Best tabletop conference phone   Luis F Urrea
• [Freeswitch-users] Audio issues when FS as PSTN/SBC Gateway for OpenSIPS   Deon Vermeulen
• [Freeswitch-users] Sangoma Media Gateway   Deon Vermeulen
• [Freeswitch-users] Sangoma Media Gateway   Deon Vermeulen
• [Freeswitch-users] event   David Villasmil
• [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to dialplan extebsion   Kozak Vladimir
• [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to dialplan extebsion   Kozak Vladimir
• [Freeswitch-users] Problem with XML CDR   Nice Voip
• [Freeswitch-users] Problem with XML CDR   Nice Voip
• [Freeswitch-users] Problem with XML CDR   Nice Voip
• [Freeswitch-users] unable to connect to freeswitch server using x-lite   Dhairya Vora
• [Freeswitch-users] Calls not established : See the error details   Dhairya Vora
• [Freeswitch-users] Calls not established : See the error details   Dhairya Vora
• [Freeswitch-users] Calls not established : See the error details   Dhairya Vora
• [Freeswitch-users] freeswitch 1.0.7 required   Dhairya Vora
• [Freeswitch-users] freeswitch 1.0.7 required   Dhairya Vora
• [Freeswitch-users] freeswitch setup error at making in lang/cmu_us_kal ...   Dhairya Vora
• [Freeswitch-users] freeswitch setup error at making in lang/cmu_us_kal ...   Dhairya Vora
• [Freeswitch-users] [Newbie] Unable to play .mp3 file   Dhairya Vora
• [Freeswitch-users] [Newbie] Unable to play .mp3 file   Dhairya Vora
• [Freeswitch-users] [Newbie] Unable to play .mp3 file   Dhairya Vora
• [Freeswitch-users] CDR   Dhairya Vora
• [Freeswitch-users] ESL: not bein able to determine when SIP gateway is down when originating a call   Bill W.
• [Freeswitch-users] ESL: not bein able to determine when SIP gateway is down when originating a call   Bill W.
• [Freeswitch-users] V92 modem with freeswitch   ankIT WALiA
• [Freeswitch-users] Setup with Bandwidth.com   Wayne
• [Freeswitch-users] Various question - CallerID / Call transfer via REFER and CDR   Weigel, Stefan
• [Freeswitch-users] Various question - CallerID / Call transfer via REFER and CDR   Weigel, Stefan
• [Freeswitch-users] Original Caller ID/number after attended transfer   Weigel, Stefan
• [Freeswitch-users] Original Caller ID/number after attended transfer   Weigel, Stefan
• [Freeswitch-users] Original Caller ID/number after attended transfer   Weigel, Stefan
• [Freeswitch-users] Original Caller ID/number after attended transfer   Weigel, Stefan
• [Freeswitch-users] Playback .wma/.wmv   Brian West
• [Freeswitch-users] event   Brian West
• [Freeswitch-users] 400 Bad Record-Route Header   Brian West
• [Freeswitch-users] SIP proxy collect DTMF using FS   Brian West
• [Freeswitch-users] incompatible destination because of crypto on incoming calls   Brian West
• [Freeswitch-users] ACL: inside out   Brian West
• [Freeswitch-users] freeswitch 1.0.7 required   Brian West
• [Freeswitch-users] FS to a Sonus SIP trunk   Brian West
• [Freeswitch-users] Discovered problem with PocketSphinx build?   Brian West
• [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip   Brian West
• [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip   Brian West
• [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip   Brian West
• [Freeswitch-users] Flowroute Users   Brian West
• [Freeswitch-users] Parsing Error on conference_set_auto_outcall   Brian West
• [Freeswitch-users] Looking up variable from config   Brian West
• [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR   Stephen Wilkey
• [Freeswitch-users] Problem on some of of the calls (not calling / no Hangupcompleted event / Not released)   Frankie Yiu
• [Freeswitch-users] Help with dual IP gateways   Lars Zeb
• [Freeswitch-users] Start FreeSWITCH as a Windows service with -nonat option   Aleš Zelenik
• [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux   Cyril Zlachevsky
• [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux   Cyril Zlachevsky
• [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux   Cyril Zlachevsky
• [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux   Cyril Zlachevsky
• [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux   Cyril Zlachevsky
• [Freeswitch-users] Exchanging IMs via RTMP   afshin afzali
• [Freeswitch-users] Flex Client Receives Periodic Interrupted Audio   afshin afzali
• [Freeswitch-users] Flex Client Receives Periodic Interrupted Audio   afshin afzali
• [Freeswitch-users] freeswitch.org is down   jay binks
• [Freeswitch-users] SIP auth challenge error   covici at ccs.covici.com
• [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed   covici at ccs.covici.com
• [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed   covici at ccs.covici.com
• [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed   covici at ccs.covici.com
• [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed   covici at ccs.covici.com
• [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed   covici at ccs.covici.com
• [Freeswitch-users] commit broke one of my trunks   covici at ccs.covici.com
• [Freeswitch-users] commit broke one of my trunks   covici at ccs.covici.com
• [Freeswitch-users] commit broke one of my trunks   covici at ccs.covici.com
• [Freeswitch-users] rewind and fast forward while playing a file   covici at ccs.covici.com
• [Freeswitch-users] rewind and fast forward while playing a file   covici at ccs.covici.com
• [Freeswitch-users] freeswitch.org is down   covici at ccs.covici.com
• [Freeswitch-users] freeswitch.org is down   covici at ccs.covici.com
• [Freeswitch-users] collect dtmf and store in channel variable   covici at ccs.covici.com
• [Freeswitch-users] any IVR example in C/C++?   covici at ccs.covici.com
• [Freeswitch-users] any IVR example in C/C++?   covici at ccs.covici.com
• [Freeswitch-users] how do I fix perl interface to streamFile?   covici at ccs.covici.com
• [Freeswitch-users] having problems building mono 2.10   covici at ccs.covici.com
• [Freeswitch-users] having problems building mono 2.10   covici at ccs.covici.com
• [Freeswitch-users] having problems building mono 2.10   covici at ccs.covici.com
• [Freeswitch-users] having problems building mono 2.10   covici at ccs.covici.com
• [Freeswitch-users] having problems building mono 2.10   covici at ccs.covici.com
• [Freeswitch-users] having problems building mono 2.10   covici at ccs.covici.com
• [Freeswitch-users] INFO: Wiki update   covici at ccs.covici.com
• [Freeswitch-users] mod_conference - 2-digit DTMF and events   covici at ccs.covici.com
• [Freeswitch-users] problem using mod_managed under linux   covici at ccs.covici.com
• [Freeswitch-users] problem using mod_managed under linux   covici at ccs.covici.com
• [Freeswitch-users] problem using mod_managed under linux   covici at ccs.covici.com
• [Freeswitch-users] problem using mod_managed under linux   covici at ccs.covici.com
• [Freeswitch-users] Last Chance for ClueCon 2011: Get Your Hotel Today   marketing at cluecon.com
• [Freeswitch-users] SIP auth challenge error   cogs66
• [Freeswitch-users] SIP auth challenge error   cogs66
• [Freeswitch-users] SIP auth challenge error   cogs66
• [Freeswitch-users] freeswitch 1.0.7 required   curriegrad2004
• [Freeswitch-users] openssl deprecated on OS X Lion and MacPorts with latest Git head?   curriegrad2004
• [Freeswitch-users] freeswitch.org is down   curriegrad2004
• [Freeswitch-users] freeswitch.org is down   curriegrad2004
• [Freeswitch-users] freeswitch.org is down   curriegrad2004
• [Freeswitch-users] Secure connection   curriegrad2004
• [Freeswitch-users] Open Source Billing for FreeSWITCH to get public soon ..   curriegrad2004
• [Freeswitch-users] Custom SIP Profile   curriegrad2004
• [Freeswitch-users] INFO: Wiki update   curriegrad2004
• [Freeswitch-users] fs_irvd   curriegrad2004
• [Freeswitch-users] "make current" does not work with Fedora 14   curriegrad2004
• [Freeswitch-users] AMR transcoding   curriegrad2004
• [Freeswitch-users] Error compling jsxml.c   daniel
• [Freeswitch-users] Returned mail: see transcript for details   johnd at defyne.org
• [Freeswitch-users] Receiving junk ESL events   dma
• [Freeswitch-users] mod_radius_cdr does not work   fieldpeak
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   fieldpeak
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   fieldpeak
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   fieldpeak
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   fieldpeak
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   fieldpeak
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   fieldpeak
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   fieldpeak
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   fieldpeak
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   fieldpeak
• [Freeswitch-users] Mod_rad_auth issue for FS working with FreeRadius server   fieldpeak
• [Freeswitch-users] Custom SIP Profile   hmkias at gmail.com
• [Freeswitch-users] TLS+SRTP= 415 Bad Security Level   vaad.fabi at gmail.com
• [Freeswitch-users] Goip GSM Gateway works great with FreeSwitch!   jcgpoza gonzalez
• [Freeswitch-users] Help with dual IP gateways   chrisbware at interfree.it
• [Freeswitch-users] Lua not playing wav files   chrisbware at interfree.it
• [Freeswitch-users] freetdm api - ftdm_channel_read()   jfabo
• [Freeswitch-users] collect dtmf and store in channel variable   vip killa
• [Freeswitch-users] collect dtmf and store in channel variable   vip killa
• [Freeswitch-users] collect dtmf and store in channel variable   vip killa
• [Freeswitch-users] collect dtmf and store in channel variable   vip killa
• [Freeswitch-users] collect dtmf and store in channel variable   vip killa
• [Freeswitch-users] collect dtmf and store in channel variable   vip killa
• [Freeswitch-users] collect dtmf and store in channel variable   vip killa
• [Freeswitch-users] collect dtmf and store in channel variable   vip killa
• [Freeswitch-users] collect dtmf and store in channel variable   vip killa
• [Freeswitch-users] enterprise deployment not working   vip killa
• [Freeswitch-users] enterprise deployment not working   vip killa
• [Freeswitch-users] enterprise deployment not working   vip killa
• [Freeswitch-users] enterprise deployment not working   vip killa
• [Freeswitch-users] enterprise deployment not working   vip killa
• [Freeswitch-users] May FS module Application NOT return data like FS module API?   king2kin
• [Freeswitch-users] Weird: Sofia-sip failed to be compiled on Visual Studio 2008 but it was OK three weeks ago   king2kin
• [Freeswitch-users] Weird: Sofia-sip failed to be compiled on Visual Studio 2008 but it was OK three weeks ago   king2kin
• [Freeswitch-users] How to start IVR script via dialplan by making an outbound call   king2kin
• [Freeswitch-users] How to start IVR script via dialplan by making an outbound call   king2kin
• [Freeswitch-users] How to start IVR script via dialplan by making an outbound call   king2kin
• [Freeswitch-users] Session ends unexpectedly during record dial plan usage   king2kin
• [Freeswitch-users] session:recordFile(-) always creates empty wav file during outbound IVR call   king2kin
• [Freeswitch-users] ptime   michael knop
• [Freeswitch-users] ptime   michael knop
• [Freeswitch-users] Freeswitch in server mode with OpenFire   krunci
• [Freeswitch-users] odbc basic_calls   mazilo
• [Freeswitch-users] ACL: inside out   micha
• [Freeswitch-users] Open Source Billing for FreeSWITCH to get public soon ..   nbhatti
• [Freeswitch-users] How to for video call ?   bino oetomo
• [Freeswitch-users] How to for video call ?   bino oetomo
• [Freeswitch-users] How to for video call ?   bino oetomo
• [Freeswitch-users] How to for video call ?   bino oetomo
• [Freeswitch-users] freetdm error message   ovvenkat
• [Freeswitch-users] freetdm error message   ovvenkat
• [Freeswitch-users] error loading mod_xml_cdr   ovvenkat
• [Freeswitch-users] error loading mod_xml_cdr   ovvenkat
• [Freeswitch-users] [Freeswitch-dev] Meetup Paris?   ovvenkat
• [Freeswitch-users] Freeswitch Hangup Event Tracking   rahulkrishna
• [Freeswitch-users] Freeswitch Hangup Event Tracking   rahulkrishna
• [Freeswitch-users] Freeswitch Hangup Event Tracking   rahulkrishna
• [Freeswitch-users] api_hangup_hook and hupall   rahulkrishna
• [Freeswitch-users] intercept sethanguphook failure   rentmycoder rentmycoder
• [Freeswitch-users] [Freeswitch-dev] Meetup Paris?   guru singh
• [Freeswitch-users] Slightly OT - Freeswitch and Vitelity?   dave.redmore at spigotnetworks.com
• [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR   steve2010
• [Freeswitch-users] Incoming DID   sucsession
• [Freeswitch-users] Incoming DID   sucsession
• [Freeswitch-users] Help setting up SIP reg   lloyd thomas
• [Freeswitch-users] Help setting up SIP reg   lloyd thomas
• [Freeswitch-users] Help setting up SIP reg   lloyd thomas
• [Freeswitch-users] Help setting up SIP reg   lloyd thomas
• [Freeswitch-users] Help setting up SIP reg   lloyd thomas
• [Freeswitch-users] Help setting up SIP reg   lloyd thomas
• [Freeswitch-users] outgoing SIP profiles setup   lloyd thomas
• [Freeswitch-users] outgoing SIP profiles setup   lloyd thomas
• [Freeswitch-users] any IVR example in C/C++?   xl127
• [Freeswitch-users] any IVR example in C/C++?   xl127
• [Freeswitch-users] any IVR example in C/C++?   xl127
• [Freeswitch-users] any IVR example in C/C++?   xl127
• [Freeswitch-users] How to do "search" in freeswitch-users Archives?   xl127
• [Freeswitch-users] mod_spidermonkey loading error   xl127
• [Freeswitch-users] mod_spidermonkey loading error   xl127
• [Freeswitch-users] mod_spidermonkey loading error   xl127
• [Freeswitch-users] mod_spidermonkey loading error   xl127
• [Freeswitch-users] "make current" does not work with Fedora 14   xl127
• [Freeswitch-users] "make current" does not work with Fedora 14   xl127
• [Freeswitch-users] regex help   babak yakhchali
• [Freeswitch-users] please help! Invalid file format [wav]   ran zhang
• [Freeswitch-users] Sofia not taking new ipaddress from local ip '127.0.0.1'   ran zhang
• [Freeswitch-users] FS terminated maybe due to memory leak?   ran zhang
• [Freeswitch-users] please help!!! playing tone only when 3rd party joins in bridging conference   ran zhang

Last message date: Wed Aug 31 23:38:51 MSD 2011
Archived on: Wed Aug 31 23:38:56 MSD 2011

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