April 2010 Archives by author

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Starting: Thu Apr 1 00:53:34 CDT 2010
Ending: Fri Apr 30 19:56:23 CDT 2010
Messages: 1231

• [asterisk-users] Converting GSM calls to SIP   William Stillwell (Lists)
• [asterisk-users] Asterisk choking on voice messages announcements   Darrick Hartman (lists)
• [asterisk-users] RTCP How to stop   nakaji at 02.246.ne.jp
• [asterisk-users] RTP over TCP   adamk at 3a.hu
• [asterisk-users] RTP over TCP   adamk at 3a.hu
• [asterisk-users] new in Bridge(), How does it work?   675842709
• [asterisk-users] Installing For AsteirskAddon   675842709
• [asterisk-users] Asterisk and Patton   A.Santoro
• [asterisk-users] Asterisk and Patton   A.Santoro
• [asterisk-users] Asterisk and Patton   A.Santoro
• [asterisk-users] Access denied for user 'a2billinguser   Daniel Abreu
• [asterisk-users] call files in 1.6   Jonathan Addleman
• [asterisk-users] FAX 2 mail configuration   Kashif Ali
• [asterisk-users] Callerid over IAX Trunks   Alyed
• [asterisk-users] Remote registering fails   Alyed
• [asterisk-users] Remote registering fails   Alyed
• [asterisk-users] Remote registering fails   Alyed
• [asterisk-users] Being attacked by an Amazon EC2 ...   Alyed
• [asterisk-users] All incoming calls landing in [customers] context   Alyed
• [asterisk-users] meetme / upgrade to 1.6.2.6   Alyed
• [asterisk-users] X-lite direct sip call - Is it possible?   Alyed
• [asterisk-users] X-lite direct sip call - Is it possible?   Alyed
• [asterisk-users] meetme / upgrade to 1.6.2.6   Alyed
• [asterisk-users] meetme / upgrade to 1.6.2.6   Alyed
• [asterisk-users] Problems with t38modem and bitrate sent to t38-termination service   Miguel Amez
• [asterisk-users] Being attacked by an Amazon EC2 ...   Roderick A. Anderson
• [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)   Goke M Aruna
• [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)   Goke M Aruna
• [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)   Goke M Aruna
• [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)   Goke M Aruna
• [asterisk-users] Cache sound files for faster processing   David Backeberg
• [asterisk-users] How to log into separate file   David Backeberg
• [asterisk-users] long return times from System() calls with 1.6.2.6?   David Backeberg
• [asterisk-users] long return times from System() calls with 1.6.2.6?   David Backeberg
• [asterisk-users] long return times from System() calls with 1.6.2.6?   David Backeberg
• [asterisk-users] asterisk-users Digest, Vol 69, Issue 16   David Backeberg
• [asterisk-users] Problems with Fax over TDM410P   David Backeberg
• [asterisk-users] res fax help   David Backeberg
• [asterisk-users] Asterisk & Timezones   David Backeberg
• [asterisk-users] cause 66 - Channel not implemented   David Backeberg
• [asterisk-users] Merge .csv files   David Backeberg
• [asterisk-users] Problems with Fax over TDM410P   David Backeberg
• [asterisk-users] Problems with Fax over TDM410P   David Backeberg
• [asterisk-users] [Conference] Audio/Video   David Backeberg
• [asterisk-users] FastAGiin Windows Server   David Backeberg
• [asterisk-users] Why app_fax.so there is no in asterisk16-1.6.2.6-1_centos5.x86_64.rpm?   David Backeberg
• [asterisk-users] long return times from System() calls with 1.6.2.6?   David Backeberg
• [asterisk-users] long return times from System() calls with 1.6.2.6?   David Backeberg
• [asterisk-users] long return times from System() calls with 1.6.2.6?   David Backeberg
• [asterisk-users] RTP over TCP   David Backeberg
• [asterisk-users] Jitter Buffer and MeetMe.   David Backeberg
• [asterisk-users] 1.6.2 - Pickup and SIP Replaces header   David Backeberg
• [asterisk-users] Jitter Buffer and MeetMe.   David Backeberg
• [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286   David Backeberg
• [asterisk-users] Asterisk 1.6, dialplans, and IVR   Greg Banschbach
• [asterisk-users] How to debug the problem of Asterisk using so much of CPU percentage...?   Greg Banschbach
• [asterisk-users] Updating Asterisk and its use with MySQL   Daniel Bareiro
• [asterisk-users] Updating Asterisk and its use with MySQL   Daniel Bareiro
• [asterisk-users] Remote registering fails   Daniel Bareiro
• [asterisk-users] Remote registering fails   Daniel Bareiro
• [asterisk-users] Security tests   Daniel Bareiro
• [asterisk-users] Security tests   Daniel Bareiro
• [asterisk-users] Being attacked by an Amazon EC2 ...   Remco Barendse
• [asterisk-users] Shorewall rate limiting rules?   Remco Barendse
• [asterisk-users] [Conference] Audio/Video   Stéphane Bauland
• [asterisk-users] [Conference] Audio/Video   Stéphane Bauland
• [asterisk-users] change IP address of AA 50?   Matt Bazan
• [asterisk-users] Asterisk & Timezones   Aldo Bergamini
• [asterisk-users] Asterisk & Timezones   Aldo Bergamini
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Gareth Blades
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Gareth Blades
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Gareth Blades
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Gareth Blades
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Gareth Blades
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Gareth Blades
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Gareth Blades
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Gareth Blades
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Gareth Blades
• [asterisk-users] Starting call recording using a dynamic feature to call a macro   Gareth Blades
• [asterisk-users] Starting call recording using a dynamic feature to call a macro   Gareth Blades
• [asterisk-users] incoming call should ring on several dahdi channels   Gareth Blades
• [asterisk-users] incoming call should ring on several dahdi channels   Gareth Blades
• [asterisk-users] Strange Invite issue   Gareth Blades
• [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286   Gareth Blades
• [asterisk-users] Need help with a pika warp asterisk appliance problem.   Rod Boileau
• [asterisk-users] OpenSIPS with Asterisk Backend   Robert Borz
• [asterisk-users] Testing a sip call through Asterisk?   Sean Brady
• [asterisk-users] On CLI SIP don't appear   Sean Brady
• [asterisk-users] Dozens of SIP NOTIFY messages with unique call ID's, and the same mailbox repeated multiple times on 1.6.2.6   Sean Brady
• [asterisk-users] Odd Issue With Polycom Phones   Sean Brady
• [asterisk-users] Odd Issue With Polycom Phones   Sean Brady
• [asterisk-users] Asterisk choking on voice messages announcements   Sean Brady
• [asterisk-users] What is needed to test issue 0013573: [Patch] Allow realtime_multi_ldap to behave like other realtime_multi functions   Sean Brady
• [asterisk-users] Changing storm-prevention behaviour in logger.conf   Remco Bressers
• [asterisk-users] Changing storm-prevention behaviour in logger.conf   Remco Bressers
• [asterisk-users] Changing storm-prevention behaviour in logger.conf   Remco Bressers
• [asterisk-users] G.729 Codec problem.   Jeff Brower
• [asterisk-users] Flood of REGISTERs - attack?   Jeff Brower
• [asterisk-users] Converting GSM calls to SIP   Jeff Brower
• [asterisk-users] Converting GSM calls to SIP   Jeff Brower
• [asterisk-users] Converting GSM calls to SIP   Jeff Brower
• [asterisk-users] Slightly OT: OMA DM Solution   Jeff Brower
• [asterisk-users] Slightly OT: OMA DM Solution   Jeff Brower
• [asterisk-users] Being attacked by an Amazon EC2 ...   Jeff Brower
• [asterisk-users] Improving audio bitrate for all callers in aconference room for a podcast   Jeff Brower
• [asterisk-users] Improving audio bitrate for all callers in a conference room for a podcast   Jeff Brower
• [asterisk-users] Asterisk choking on voice messagesannouncements   Jeff Brower
• [asterisk-users] Converting GSM calls to SIP   Pascal Bruno
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Pascal Bruno
• [asterisk-users] Being attacked by an Amazon EC2 ...   Frank Bulk
• [asterisk-users] Being attacked by an Amazon EC2 ...   Frank Bulk
• [asterisk-users] Being attacked by an Amazon EC2 ...   Frank Bulk
• [asterisk-users] A matter of context   Ryan Bullock
• [asterisk-users] Asterisk choking on voice messages announcements   Ryan Bullock
• [asterisk-users] Asterisk choking on voice messages announcements   Ryan Bullock
• [asterisk-users] Follow-me to my answering machine :-(   Ryan Bullock
• [asterisk-users] Hangup after n seconds using originate ?   Ryan Bullock
• [asterisk-users] More efficient dial plan for a list of selective inbound numbers   Ryan Bullock
• [asterisk-users] Follow-me to my answering machine :-(   Ryan Bullock
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Ryan Bullock
• [asterisk-users] Dial plan question.   Ryan Bullock
• [asterisk-users] Possible AGI bug?   Leo Burd
• [asterisk-users] Possible AGI bug?   Leo Burd
• [asterisk-users] What's are the possible return values of AMI Originate when Async is set to 0?   Leo Burd
• [asterisk-users] Gateway E1 <=> Asterisk ?   Olivier CALVANO
• [asterisk-users] Agent Callback methods?   Emanuele Carbone
• [asterisk-users] asterisk running @ 100% load doing nothing   Kelvin Chan
• [asterisk-users] asterisk running @ 100% load doing nothing   Kelvin Chan
• [asterisk-users] asterisk running @ 100% load doing nothing   Kelvin Chan
• [asterisk-users] asterisk running @ 100% load doing nothing   Kelvin Chan
• [asterisk-users] SIP registration failure stops all SIP activity   Carlos Chavez
• [asterisk-users] Recording music in Queue   Carlos Chavez
• [asterisk-users] How can I record the conversations in a conference call?   Carlos Chavez
• [asterisk-users] meetme / upgrade to 1.6.2.6   Carlos Chavez
• [asterisk-users] How to record a call in a single file when transfered...   Carlos Chavez
• [asterisk-users] How to record a call in a single file when transfered...   Carlos Chavez
• [asterisk-users] How to record a call in a single file when transfered...   Carlos Chavez
• [asterisk-users] Asterisk choking on voice messages announcements   Carlos Chavez
• [asterisk-users] automatic call with call files   Adolphe Cher-Aime
• [asterisk-users] Please sign Petition - Stop Child Labour   Sarfaraz Chougule
• [asterisk-users] Asterisk and Archlinux   Christian
• [asterisk-users] Do AMI Events have timestamps?   Frank Church
• [asterisk-users] Asterisk Load Balancing with Redfone/TDMoE driver   Jorge Churio
• [asterisk-users] Linksys/Sipura SPA-3201 FXO/FSA with Asterisk   Seann Clark
• [asterisk-users] Linksys/Sipura SPA-3201 FXO/FSA with Asterisk   Seann Clark
• [asterisk-users] hardware clock drift and CDR   Seann Clark
• [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1   Seann Clark
• [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1   Seann Clark
• [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1   Seann Clark
• [asterisk-users] Strange Error -- ASterisk 1.6   Seann Clark
• [asterisk-users] Strange Error -- ASterisk 1.6   Seann Clark
• [asterisk-users] Testing a sip call through Asterisk?   Nathan Clemons
• [asterisk-users] Testing a sip call through Asterisk?   Nathan Clemons
• [asterisk-users] RTP over TCP   Nathan Clemons
• [asterisk-users] spool directories and filename   Ricardo Coelho
• [asterisk-users] spool directories and filename   Ricardo Coelho
• [asterisk-users] Merge master.csv files   Ricardo Coelho
• [asterisk-users] Merge .csv files   Ricardo Coelho
• [asterisk-users] G729 exhaustion conditions   Harel Cohen
• [asterisk-users] Call-Waiting, implementation ideas   Harel Cohen
• [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8   Tzafrir Cohen
• [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8   Tzafrir Cohen
• [asterisk-users] How set debug file for RxFax application   Tzafrir Cohen
• [asterisk-users] asterisk start with php   Tzafrir Cohen
• [asterisk-users] asterisk start with php   Tzafrir Cohen
• [asterisk-users] dial extension and play sound file from shell on asterisk server?   Tzafrir Cohen
• [asterisk-users] Please sign Petition - Stop Child Labour   Tzafrir Cohen
• [asterisk-users] Asterisk in Debian/Lenny without Junghanns.net support?   Tzafrir Cohen
• [asterisk-users] Dahdi, junghanns and qozap   Tzafrir Cohen
• [asterisk-users] Being attacked by an Amazon EC2 ...   Tzafrir Cohen
• [asterisk-users] Being attacked by an Amazon EC2   Tzafrir Cohen
• [asterisk-users] problem of "when memory become 50% or more then sound become noisy?"   Tzafrir Cohen
• [asterisk-users] cat /proc/zaptel/*   Tzafrir Cohen
• [asterisk-users] dahdi_scan and OctoBRI. Bug or feature ?   Tzafrir Cohen
• [asterisk-users] Being attacked by an Amazon EC2 ...   Tzafrir Cohen
• [asterisk-users] Merge master.csv files   Tzafrir Cohen
• [asterisk-users] DAHDI-Linux and DAHDI-Tools 2.3.0 Released   Tzafrir Cohen
• [asterisk-users] 1.6.0 verses 1.6.2   Tzafrir Cohen
• [asterisk-users] Amazon EC2 SIP floods - you can help   Tzafrir Cohen
• [asterisk-users] incoming ghost call   Tzafrir Cohen
• [asterisk-users] Manager events & safety   Tzafrir Cohen
• [asterisk-users] DAHDI Congestion cause 34   Tzafrir Cohen
• [asterisk-users] BN8S0, dahdi, wcb4xxp   Tzafrir Cohen
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Tzafrir Cohen
• [asterisk-users] Full transfer details on inbound calls   Nic Colledge
• [asterisk-users] Sending RTP media to a different server than SIP Signaling   Joshua Colp
• [asterisk-users] 1.6.0 verses 1.6.2   Andrea Cristofanini
• [asterisk-users] RPID on called party   CunningPike
• [asterisk-users] Confusion on call forwarding   Klaus Darilion
• [asterisk-users] PRI - Native ZAP bridge fails - Is this my patch?   Klaus Darilion
• [asterisk-users] DAHDI Congestion cause 34   Chris Datfung
• [asterisk-users] DAHDI Congestion cause 34   Chris Datfung
• [asterisk-users] Dropped Calls   Brent Davidson
• [asterisk-users] Transfer_CONTEXT behaviour   Steve Davies
• [asterisk-users] Transfer_CONTEXT behaviour   Steve Davies
• [asterisk-users] SIP devide call-forward behaviour and CDRs   Steve Davies
• [asterisk-users] Transfer_CONTEXT behaviour   Steve Davies
• [asterisk-users] SIP devide call-forward behaviour and CDRs   Steve Davies
• [asterisk-users] SIP authentication   Steve Davies
• [asterisk-users] Improving audio bitrate for all callers in a conference room for a podcast   Patrick Davila
• [asterisk-users] Improving audio bitrate for all callers in a conference room for a podcast   Patrick Davila
• [asterisk-users] Improving audio bitrate for all callers in a conference room for a podcast   Patrick Davila
• [asterisk-users] OT: Wireless headset / phone combination   Alec Davis
• [asterisk-users] OT: Wireless headset / phone combination   Alec Davis
• [asterisk-users] polarity reverse   Alec Davis
• [asterisk-users] polarity reverse   Alec Davis
• [asterisk-users] polarity reverse   Alec Davis
• [asterisk-users] run script after completed   Necati Demir
• [asterisk-users] Safe_asterisk doesn't exists???   Danny Dias
• [asterisk-users] Problems with Fax over TDM410P   Danny Dias
• [asterisk-users] Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCodes   Danny Dias
• [asterisk-users] Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCode   Danny Dias
• [asterisk-users] Problems with Fax over TDM410P   Danny Dias
• [asterisk-users] Asterisk DIES with no trace. PLEASE   Danny Dias
• [asterisk-users] Problems with Fax over TDM410P   Danny Dias
• [asterisk-users] How to set up Fax on Asterisk - Using analog Fax machines and HT502 (or FXS of a Digium TDM410P)   Danny Dias
• [asterisk-users] incoming ghost call   Danny Dias
• [asterisk-users] IBM X3650 with Asterisk???   Danny Dias
• [asterisk-users] How can I record the conversations in a conference call?   Juan David Diaz
• [asterisk-users] How can I record the conversations in a conference call?   Juan David Diaz
• [asterisk-users] Asterisk Query   Juan David Diaz
• [asterisk-users] G.729 Codec problem.   Jim Dickenson
• [asterisk-users] Delay the HungUp   Jim Dickenson
• [asterisk-users] IAX connection slow between 1.4 and 1.6 dists   Jim Dickenson
• [asterisk-users] Put a call on hold with Manager   Jim Dickenson
• [asterisk-users] Read Timeout   Jim Dickenson
• [asterisk-users] Hangup after n seconds using originate ?   Jim Dickenson
• [asterisk-users] Playback all the sound files   Jim Dickenson
• [asterisk-users] dialplan question   Jim Dickenson
• [asterisk-users] dialplan question   Jim Dickenson
• [asterisk-users] Dial plan question.   Jim Dickenson
• [asterisk-users] dialplan   Jim Dickenson
• [asterisk-users] Dial plan question.   Jim Dickenson
• [asterisk-users] Issue with (pattern) matching extension   Jim Dickenson
• [asterisk-users] All incoming calls landing in [customers] context   Mike Diehl
• [asterisk-users] Asterisk and Patton   Carlo Dimaggio
• [asterisk-users] Asterisk and Patton   Carlo Dimaggio
• [asterisk-users] Calls drop after 20 seconds   Doug
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   Edo
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   Edo
• [asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution   Steve Edwards
• [asterisk-users] asterisk start with php   Steve Edwards
• [asterisk-users] asterisk start with php   Steve Edwards
• [asterisk-users] asterisk start with php   Steve Edwards
• [asterisk-users] asterisk start with php   Steve Edwards
• [asterisk-users] spool directories and filename   Steve Edwards
• [asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution   Steve Edwards
• [asterisk-users] spool directories and filename   Steve Edwards
• [asterisk-users] Cache sound files for faster processing   Steve Edwards
• [asterisk-users] Limit Number Of Simultaneous Outbound SIP Calls   Steve Edwards
• [asterisk-users] IAX Problem   Steve Edwards
• [asterisk-users] IAX Call Rejected (was IAX Problem)   Steve Edwards
• [asterisk-users] PSTN issues   Steve Edwards
• [asterisk-users] How to log into separate file   Steve Edwards
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   Steve Edwards
• [asterisk-users] IVR menu sound processing for AMR and GSM + live test available   Steve Edwards
• [asterisk-users] softphone help   Steve Edwards
• [asterisk-users] Merge master.csv files   Steve Edwards
• [asterisk-users] Possible AGI bug?   Steve Edwards
• [asterisk-users] Conference Meetme   Steve Edwards
• [asterisk-users] FastAGiin Windows Server   Steve Edwards
• [asterisk-users] Interpbx connection   Steve Edwards
• [asterisk-users] Voice mail "maxmessage " setting per mail box   Steve Edwards
• [asterisk-users] Voice mail "maxmessage " setting per mail box   Steve Edwards
• [asterisk-users] 1.6.2 No "soft hangup"?   Steve Edwards
• [asterisk-users] 1.6.2 No "soft hangup"?   Steve Edwards
• [asterisk-users] Asterisk choking on voice messages announcements   Steve Edwards
• [asterisk-users] Time difference in CSV CDR's and MySQL CDR's   Steve Edwards
• [asterisk-users] Playback all the sound files   Steve Edwards
• [asterisk-users] hardware clock drift and CDR   Steve Edwards
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Steve Edwards
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Steve Edwards
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Steve Edwards
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Steve Edwards
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Steve Edwards
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Steve Edwards
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Steve Edwards
• [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8   Lenz Emilitri
• [asterisk-users] Evaluating Asterisk   Lenz Emilitri
• [asterisk-users] Queue call to specific queuemember   Lenz Emilitri
• [asterisk-users] Split E1 ISDN service for another device.   voip88 Eric
• [asterisk-users] D-Channel Span Up without Down   voip88 Eric
• [asterisk-users] Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?]   Jose P. Espinal
• [asterisk-users] Asterisk room monitor   C F
• [asterisk-users] Continuing after a TIMEOUT(absolute)   C F
• [asterisk-users] Call-Waiting, implementation ideas   C F
• [asterisk-users] Asterisk/Polycom Dialed Party Name   Karl Fife
• [asterisk-users] Asterisk/Polycom Dialed Party Name   Karl Fife
• [asterisk-users] asterisk-gplonly dependency in asterisk-addons RPM   Kevin P. Fleming
• [asterisk-users] RTCP How to stop   Kevin P. Fleming
• [asterisk-users] trying app_fax.c   Kevin P. Fleming
• [asterisk-users] Split E1 ISDN service for another device.   Kevin P. Fleming
• [asterisk-users] asterisk-gplonly dependency in asterisk-addons RPM   Kevin P. Fleming
• [asterisk-users] long return times from System() calls with 1.6.2.6?   Kevin P. Fleming
• [asterisk-users] Need help with a pika warp asterisk appliance problem.   Kevin P. Fleming
• [asterisk-users] Is svn.asterisk.org down ?   Kevin P. Fleming
• [asterisk-users] Asterisk/Polycom Dialed Party Name   Kevin P. Fleming
• [asterisk-users] How to set up Fax on Asterisk - Using analog Fax machines and HT502 (or FXS of a Digium TDM410P)   Kevin P. Fleming
• [asterisk-users] How to do analog e&m on asterisk?   Kevin P. Fleming
• [asterisk-users] How to do analog e&m on asterisk?   Kevin P. Fleming
• [asterisk-users] Dahdi will not compile on Unbuntu Studio Linux 9.10 (Karmic) 32bit   Kevin P. Fleming
• [asterisk-users] Dahdi will not compile on Unbuntu Studio Linux 9.10 (Karmic) 32bit   Kevin P. Fleming
• [asterisk-users] Problems for Skype for Asterisk   Kevin P. Fleming
• [asterisk-users] No change in payload. (SDP)   Kevin P. Fleming
• [asterisk-users] OfficeSIP Communications Makes Its VoIP SIP Products Open Source   Vitali Fomine
• [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8   Vitali Fomine
• [asterisk-users] Evaluating Asterisk   Ted Foote
• [asterisk-users] Anyone coming to Paris next week for AstriEurope?   Matthew Fredrickson
• [asterisk-users] t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED   Matthew Fredrickson
• [asterisk-users] Agent Callback methods?   Joe Freeman
• [asterisk-users] res fax help   Joe Freeman
• [asterisk-users] Rebooting Polycom's - Could not create address for 'XXXX'   Jose Flores Galicia
• [asterisk-users] Linksys/Sipura SPA-3201 FXO/FSA with Asterisk   Jose Flores Galicia
• [asterisk-users] Linksys/Sipura SPA-3201 FXO/FSA with Asterisk   Jose Flores Galicia
• [asterisk-users] Swaping out phones.   Jose Flores Galicia
• [asterisk-users] More efficient dial plan for a list of selective inbound numbers   Jian Gao
• [asterisk-users] Playback all the sound files   Jian Gao
• [asterisk-users] Playback all the sound files   Jian Gao
• [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286   Jian Gao
• [asterisk-users] FAX 2 mail configuration   Tiago Geada
• [asterisk-users] trying app_fax.c   Jerry Geis
• [asterisk-users] call files in 1.6   Jerry Geis
• [asterisk-users] call files in 1.6   Jerry Geis
• [asterisk-users] tones detection   Jerry Geis
• [asterisk-users] B400P and A1200P changes card order   Peter Gelencser
• [asterisk-users] incoming call should ring on several dahdi channels   Peter Gelencser
• [asterisk-users] B400P and A1200P changes card order   Peter Gelencser
• [asterisk-users] B400P card crashes conncection   Peter Gelencser
• [asterisk-users] AMD reporting NOTSURE most of the time   Chris Gentle
• [asterisk-users] Monitoring calls via sound card   Chris Gentle
• [asterisk-users] AMD reporting NOTSURE most of the time   Chris Gentle
• [asterisk-users] callprogress issue   Chris Gentle
• [asterisk-users] Anyone coming to Paris next week for AstriEurope?   Georghy
• [asterisk-users] AGI + Dial + stream file ?   Godson Gera
• [asterisk-users] dial extension and play sound file from shell on asterisk server?   Godson Gera
• [asterisk-users] problem of registration with Asterisk using exosip2   Idriss Ghodhbane
• [asterisk-users] Continuous bothering message -- Remote UNIX connection disconnected   David Gibbons
• [asterisk-users] Changing storm-prevention behaviour in logger.conf   Lyle Giese
• [asterisk-users] SpiderMux?   Lyle Giese
• [asterisk-users] Dahdi will not compile on Unbuntu Studio Linux 9.10 (Karmic) 32bit   Steve Gladden
• [asterisk-users] Dahdi will not compile on Unbuntu Studio Linux 9.10 (Karmic) 32bit   Steve Gladden
• [asterisk-users] Building Asterisk-RPM for 1.4.24.1   Thorolf Godawa
• [asterisk-users] FastAGiin Windows Server   Jimmy Godbout
• [asterisk-users] PSTN issues   Flavio Goncalves
• [asterisk-users] scratchy sound   Flavio Goncalves
• [asterisk-users] RES: Cache sound files for faster processing   Flavio E. Goncalves
• [asterisk-users] OT: Wireless headset / phone combination   Michael Graves
• [asterisk-users] RTP over TCP   Michael Graves
• [asterisk-users] RTP Timeouts not clearing calls   Paddy Grice
• [asterisk-users] VOIP Monitoring tools........   Paddy Grice
• [asterisk-users] High Availability - Shared Database - Ideas?   Robert Grignon
• [asterisk-users] polarity reverse   Justas Gulbinskas
• [asterisk-users] polarity reverse   Justas Gulbinskas
• [asterisk-users] polarity reverse   Justas Gulbinskas
• [asterisk-users] E3 Card on Asterisk ?   Anita Hall
• [asterisk-users] canary_thread   Darrick Hartman
• [asterisk-users] Being attacked by an Amazon EC2 ...   Darrick Hartman
• [asterisk-users] Being attacked by an Amazon EC2 ...   Darrick Hartman
• [asterisk-users] Flood of REGISTERs - attack?   Chris Hastie
• [asterisk-users] Flood of REGISTERs - attack?   Chris Hastie
• [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8   Gordon Henderson
• [asterisk-users] OT: Wireless headset / phone combination   Gordon Henderson
• [asterisk-users] Being attacked by an Amazon EC2 ...   Gordon Henderson
• [asterisk-users] Being attacked by an Amazon EC2 ...   Gordon Henderson
• [asterisk-users] Being attacked by an Amazon EC2 ...   Gordon Henderson
• [asterisk-users] Being attacked by an Amazon EC2 ...   Gordon Henderson
• [asterisk-users] Being attacked by an Amazon EC2 ...   Gordon Henderson
• [asterisk-users] iptables miss up phone calls if not used properly   Gordon Henderson
• [asterisk-users] DAHDI-Linux and DAHDI-Tools 2.3.0 Released   Gordon Henderson
• [asterisk-users] Spam and that recent 'attack' ...   Gordon Henderson
• [asterisk-users] Yesterday EC2, today Netnation Europe V.O.F.   Gordon Henderson
• [asterisk-users] Being attacked by an Amazon EC2 ...   Gordon Henderson
• [asterisk-users] Security tests   Gordon Henderson
• [asterisk-users] VoIP monitoring tools   Gordon Henderson
• [asterisk-users] hardware clock drift and CDR   Gordon Henderson
• [asterisk-users] hardware clock drift and CDR   Gordon Henderson
• [asterisk-users] hardware clock drift and CDR   Gordon Henderson
• [asterisk-users] What is needed to test issue 0013573: [Patch] Allow realtime_multi_ldap to behave like other realtime_multi functions   Gavin Henry
• [asterisk-users] How Cisco ATA 186 through SCCP with skinny.conf ?!   Tamer Higazi
• [asterisk-users] Asterisk load balancing and failover   Ngo-Vi Hoai-Anh
• [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)   Ngo-Vi Hoai-Anh
• [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)   Ngo-Vi Hoai-Anh
• [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)   Ngo-Vi Hoai-Anh
• [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286   Dean Hoover
• [asterisk-users] asterisk start with php   Steve Howes
• [asterisk-users] Being attacked by an Amazon EC2 ...   Steve Howes
• [asterisk-users] cause 66 - Channel not implemented   Steve Howes
• [asterisk-users] Is svn.asterisk.org down ?   Steve Howes
• [asterisk-users] Delay the HungUp   Steve Howes
• [asterisk-users] Realtime changes not reflected realtime   Steve Howes
• [asterisk-users] Help with FastAGI server in Windows   Steve Howes
• [asterisk-users] Asterisk choking on voice messages announcements   Steve Howes
• [asterisk-users] Dial plan question.   Steve Howes
• [asterisk-users] Issue with (pattern) matching extension   Steve Howes
• [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286   Steve Howes
• [asterisk-users] Asterisk room monitor   Mark Hulber
• [asterisk-users] Asterisk room monitor   Mark Hulber
• [asterisk-users] [OpenSIPS-Users] OpenSIPS with Asterisk Backend   Bogdan-Andrei Iancu
• [asterisk-users] shared lines (sla) with Asterisk 1.4.26, any hints?   Giorgio Incantalupo
• [asterisk-users] Necessary hardware   Ioan Indreias
• [asterisk-users] Problem with Sangoma A104 and euroisdn pri   Ioan Indreias
• [asterisk-users] Asterisk room monitor   Ioan Indreias
• [asterisk-users] [Conference] Audio/Video   Ioan Indreias
• [asterisk-users] Detect if a Number is up or not   Ioan Indreias
• [asterisk-users] Follow-me to my answering machine :-(   Bryan Jacobs
• [asterisk-users] Follow-me to my answering machine :-(   Bryan Jacobs
• [asterisk-users] Follow-me to my answering machine :-(   Bryan Jacobs
• [asterisk-users] Follow-me to my answering machine :-(   Bryan Jacobs
• [asterisk-users] Follow-me to my answering machine :-(   Bryan Jacobs
• [asterisk-users] Follow-me to my answering machine :-(   Bryan Jacobs
• [asterisk-users] Follow-me to my answering machine :-(   Bryan Jacobs
• [asterisk-users] Follow-me to my answering machine :-(   Motiejus Jakštys
• [asterisk-users] asterisk running @ 100% load doing nothing   Motiejus Jakštys
• [asterisk-users] Manager events & safety   Motiejus Jakštys
• [asterisk-users] Asterisk and Archlinux   Motiejus Jakštys
• [asterisk-users] VoIP monitoring tools   Motiejus Jakštys
• [asterisk-users] Detect if a Number is up or not   Motiejus Jakštys
• [asterisk-users] Record call without caller interference   Motiejus Jakštys
• [asterisk-users] E3 Card on Asterisk ?   Motiejus Jakštys
• [asterisk-users] dialplan   Motiejus Jakštys
• [asterisk-users] Asterisk stopping for no reason   Motiejus Jakštys
• [asterisk-users] there have any one run asterisk on ubuntu enterprise cloud ?   Jeffery
• [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8   Tommy Botten Jensen
• [asterisk-users] Queue issues   Tommy Botten Jensen
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Per Jessen
• [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime   Per Jessen
• [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8   Olle E. Johansson
• [asterisk-users] Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?]   Olle E. Johansson
• [asterisk-users] Minimalize jitter in VoIP calls   John
• [asterisk-users] 1.6.2 No "soft hangup"?   Steve Johnson
• [asterisk-users] Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?]   Steve Johnson
• [asterisk-users] Read Timeout   Dan Journo
• [asterisk-users] Read Timeout   Dan Journo
• [asterisk-users] play a sound from the callee before putting it in connection.   Dan Journo
• [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286   Dan Journo
• [asterisk-users] Calls Dropping   Dan Journo
• [asterisk-users] SNOM M9 : expand range   Jonas Kellens
• [asterisk-users] SNOM M9 base station A to base station B   Jonas Kellens
• [asterisk-users] Realtime changes not reflected realtime   Jonas Kellens
• [asterisk-users] Realtime changes not reflected realtime   Jonas Kellens
• [asterisk-users] Realtime changes not reflected realtime   Jonas Kellens
• [asterisk-users] Realtime changes not reflected realtime   Jonas Kellens
• [asterisk-users] Portech MV-374 does not register   Jonas Kellens
• [asterisk-users] Portech MV-374 does not register   Jonas Kellens
• [asterisk-users] Portech MV-374 does not register   Jonas Kellens
• [asterisk-users] Portech MV-374 does not register   Jonas Kellens
• [asterisk-users] Portech MV-374 does not register   Jonas Kellens
• [asterisk-users] Portech MV-374 does not register behind NAT   Jonas Kellens
• [asterisk-users] Portech MV-374 does not register behind NAT   Jonas Kellens
• [asterisk-users] Portech MV-374 does not register behind NAT   Jonas Kellens
• [asterisk-users] Portech MV-374 does not register behind NAT   Jonas Kellens
• [asterisk-users] Realtime changes not reflected realtime   Jonas Kellens
• [asterisk-users] Manager events & safety   Jonas Kellens
• [asterisk-users] Record call without caller interference   Jonas Kellens
• [asterisk-users] Execute Macro when queue is answered   Jonas Kellens
• [asterisk-users] PRI Gurus ONLY - Too complex of an issue   Don Kelly
• [asterisk-users] PRI Gurus ONLY - Too complex of an issue   Don Kelly
• [asterisk-users] Follow-me to my answering machine :-(   Don Kelly
• [asterisk-users] asterisk-gplonly dependency in asterisk-addons RPM   Richard Kenner
• [asterisk-users] testexpr2   Richard Kenner
• [asterisk-users] testexpr2   Richard Kenner
• [asterisk-users] 'o' option on Dial application   Richard Kenner
• [asterisk-users] Problems for Skype for Asterisk   Richard Kenner
• [asterisk-users] Problems for Skype for Asterisk   Richard Kenner
• [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1   Richard Kenner
• [asterisk-users] E3 Card on Asterisk ?   Bill Kenworthy
• [asterisk-users] Access denied for user 'a2billinguser   Shariq Khan
• [asterisk-users] Ring Two Extensions Simultaneously with different caller ID values?   Lincoln King-Cliby
• [asterisk-users] Ring Two Extensions Simultaneously with different caller ID values? [SOLVED]   Lincoln King-Cliby
• [asterisk-users] Dynamic agent showing as "Invalid"   Jordan Kirby
• [asterisk-users] Split E1 ISDN service for another device.   Klaverstyn, David C
• [asterisk-users] Follow-me to my answering machine :-(   Barry L. Kline
• [asterisk-users] Follow-me to my answering machine :-(   Barry L. Kline
• [asterisk-users] Being attacked by an Amazon EC2 ...   Philipp von Klitzing
• [asterisk-users] Being attacked by an Amazon EC2 ...   Philipp von Klitzing
• [asterisk-users] Full transfer details on inbound calls   Philipp von Klitzing
• [asterisk-users] Using chan_lcr (and mISDN v2) ?   Philipp von Klitzing
• [asterisk-users] Conference Meetme   Philipp von Klitzing
• [asterisk-users] How can I record the conversations in a conference call?   Philipp von Klitzing
• [asterisk-users] Delay the HungUp   Philipp von Klitzing
• [asterisk-users] DIALSTATUS variable and qualify=no   Philipp von Klitzing
• [asterisk-users] Improving audio bitrate for all callers in a conference room for a podcast   Philipp von Klitzing
• [asterisk-users] Security tests   Philipp von Klitzing
• [asterisk-users] RTP over TCP   Philipp von Klitzing
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Philipp von Klitzing
• [asterisk-users] Asterisk and Patton   Philipp von Klitzing
• [asterisk-users] Asterisk and Patton   Philipp von Klitzing
• [asterisk-users] Need to patch Asterisk for problem with FreePBX Call Confirmation   Matthew A Kolberg
• [asterisk-users] Need to patch Asterisk for problem with FreePBX Call Confirmation   Matthew A Kolberg
• [asterisk-users] Necessary hardware   Kosa
• [asterisk-users] DIALSTATUS variable and qualify=no   Rustam Kovhaev
• [asterisk-users] DAHDI User-User information "Message longer than it should be??"   Alexandr Krylovskiy
• [asterisk-users] HDLC Receiver overrun on Wildcard TE410P   Łukasz Krzyżak
• [asterisk-users] Supporting addressing formats and unsolicited Notify   Aditya Kumar
• [asterisk-users] Dial plan question.   Aditya Kumar
• [asterisk-users] Dial plan question.   Aditya Kumar
• [asterisk-users] Dial plan question.   Aditya Kumar
• [asterisk-users] Dial plan question.   Aditya Kumar
• [asterisk-users] Dial plan question.   Aditya Kumar
• [asterisk-users] No change in payload. (SDP)   Aditya Kumar
• [asterisk-users] No change in payload. (SDP)   Aditya Kumar
• [asterisk-users] Being attacked by an Amazon EC2 ...   Erik L
• [asterisk-users] zapg723toslin did not update samples   Christian Hiller - Baig Tel LTD
• [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8   Jeff LaCoursiere
• [asterisk-users] realtime jitter/latency measurements   Jeff LaCoursiere
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   Jeff LaCoursiere
• [asterisk-users] jitterbuffer   Jeff LaCoursiere
• [asterisk-users] jitterbuffer   Jeff LaCoursiere
• [asterisk-users] How can I record the conversations in a conference call?   Jeff LaCoursiere
• [asterisk-users] Testing a sip call through Asterisk?   Jeff LaCoursiere
• [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286   Jeff LaCoursiere
• [asterisk-users] Swaping out phones.   Tony LaMear
• [asterisk-users] Polycom 330 not connecting   Tony LaMear
• [asterisk-users] Exceptionally long voice queue length errors...   James Lamanna
• [asterisk-users] Problem with Sangoma A104 and euroisdn pri   James Lamanna
• [asterisk-users] Busy(20) returns non-zero and exits immediately on IAX channel   James Lamanna
• [asterisk-users] Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCodes   James Lamanna
• [asterisk-users] Repeated: Got SIP response 489 "Bad event" back from   James Lamanna
• [asterisk-users] Sending RTP media to a different server than SIP Signaling   James Lamanna
• [asterisk-users] tones detection   James Lamanna
• [asterisk-users] Asterisk + DRBD Performance   James Lamanna
• [asterisk-users] [PATCH] Make Queue announcements more consistent (1.4.26.2)   James Lamanna
• [asterisk-users] Duplicated DTMF with bridged IAX channels maybe?   James Lamanna
• [asterisk-users] Duplicated DTMF with bridged IAX channels maybe?   James Lamanna
• [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286   Geoff Lane
• [asterisk-users] canary_thread   Andrew Latham
• [asterisk-users] SNOM M9 base station A to base station B   Andrew Latham
• [asterisk-users] Issue with (pattern) matching extension   Andrew Latham
• [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8   Tilghman Lesher
• [asterisk-users] problem compiling asterisk with cdr_odbc   Tilghman Lesher
• [asterisk-users] Asterisk a-law header missing?   Tilghman Lesher
• [asterisk-users] call files in 1.6   Tilghman Lesher
• [asterisk-users] Which rule for Asterisk to Asterisk-addons compatibility ?   Tilghman Lesher
• [asterisk-users] Asterisk and MWI with Exchange 2010   Tilghman Lesher
• [asterisk-users] testexpr2   Tilghman Lesher
• [asterisk-users] testexpr2   Tilghman Lesher
• [asterisk-users] Time variables in system application   Tilghman Lesher
• [asterisk-users] Time variables in system application   Tilghman Lesher
• [asterisk-users] 'o' option on Dial application   Tilghman Lesher
• [asterisk-users] Changing storm-prevention behaviour in logger.conf   Tilghman Lesher
• [asterisk-users] Bug or feature: cdr_odbc.conf.sample   Tilghman Lesher
• [asterisk-users] 1.6.2 No "soft hangup"?   Tilghman Lesher
• [asterisk-users] 1.6.2 No "soft hangup"?   Tilghman Lesher
• [asterisk-users] Unable to load cdr_adaptive_odbc.so   Tilghman Lesher
• [asterisk-users] Unable to load cdr_adaptive_odbc.so   Tilghman Lesher
• [asterisk-users] Does 'file' command work with asterisk genereted alaw file   Tilghman Lesher
• [asterisk-users] 1.6.1.18 : app_voicemail is calling sendmail without any argument   Tilghman Lesher
• [asterisk-users] Issue with (pattern) matching extension   Tilghman Lesher
• [asterisk-users] Issue with (pattern) matching extension   Tilghman Lesher
• [asterisk-users] Need help with a pika warp asterisk appliance problem.   Timothy C Litwiller
• [asterisk-users] over running my did's   Timothy C Litwiller
• [asterisk-users] Callerid over IAX Trunks   Ye Liu
• [asterisk-users] Callerid over IAX Trunks   Ye Liu
• [asterisk-users] say.conf implementation of Indian Languages to play numbers and dates   Amit Patkar | Avhan Technologies Pvt. Ltd.
• [asterisk-users] Dial an extension with follow me   Anahi Ludueña
• [asterisk-users] Cache sound files for faster processing   Luki
• [asterisk-users] Ring Two Extensions Simultaneously with different caller ID values?   Luki
• [asterisk-users] How can I record the conversations in a conference call?   Luki
• [asterisk-users] How can I record the conversations in a conference call?   Luki
• [asterisk-users] AGI <==> DeadAGI   Luki
• [asterisk-users] SIP equivalent of zap "c" option   Julian Lyndon-Smith
• [asterisk-users] Evaluating Asterisk   Julian Lyndon-Smith
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   Doug Lytle
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   Doug Lytle
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   Doug Lytle
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   Doug Lytle
• [asterisk-users] Is restart of span a concern on PRI?   Doug Lytle
• [asterisk-users] Zap PRI failed with Cause 34 - Where to check for problems?   Doug Lytle
• [asterisk-users] AGI + Dial + stream file ?   Mickael MONSIEUR
• [asterisk-users] AGI + Dial + stream file ?   Mickael MONSIEUR
• [asterisk-users] play a sound from the callee before putting it in connection.   Mickael MONSIEUR
• [asterisk-users] play a sound from the callee before putting it in connection.   Mickael MONSIEUR
• [asterisk-users] Exceptionally long voice queue length errors...   Leif Madsen
• [asterisk-users] Asterisk and MWI with Exchange 2010   Leif Madsen
• [asterisk-users] Evaluating Asterisk   Leif Madsen
• [asterisk-users] Extensions Reload | Asterisk Freezes ? 1.4   Leif Madsen
• [asterisk-users] 1.6.2 No "soft hangup"?   Leif Madsen
• [asterisk-users] How to record a call in a single file when transfered...   Leif Madsen
• [asterisk-users] How to record a call in a single file when transfered...   Leif Madsen
• [asterisk-users] Manipulating audio in asterisk   Leif Madsen
• [asterisk-users] How to record a call in a single file when transfered...   Leif Madsen
• [asterisk-users] More efficient dial plan for a list of selective inbound numbers   Leif Madsen
• [asterisk-users] Issue with (pattern) matching extension   Leif Madsen
• [asterisk-users] Issue with (pattern) matching extension   Leif Madsen
• [asterisk-users] Issue with (pattern) matching extension   Leif Madsen
• [asterisk-users] Full transfer details on inbound calls   Ishfaq Malik
• [asterisk-users] A matter of context   Ishfaq Malik
• [asterisk-users] Calls drop after 20 seconds   Ishfaq Malik
• [asterisk-users] Record call without caller interference   Ishfaq Malik
• [asterisk-users] Issue with (pattern) matching extension   Ishfaq Malik
• [asterisk-users] AGI <==> DeadAGI   Ishfaq Malik
• [asterisk-users] Queue call to specific queuemember   Asterisk Maniac
• [asterisk-users] Asterisk hangup all outging calls after 32 seconds   Ing CIP. Alejandro Celi Mariátegui
• [asterisk-users] DTMF from SIP phone to FXS/FXO   Andres Marquez
• [asterisk-users] Repeated: Got SIP response 489 "Bad event" back from   Adrian Marsh
• [asterisk-users] Repeated: Got SIP response 489 "Bad event"back from   Adrian Marsh
• [asterisk-users] Please sign Petition - Stop Child Labour   Martin
• [asterisk-users] Being attacked by an Amazon EC2 ...   Martin
• [asterisk-users] Sending SMS problems.   Agazzini Maurizio
• [asterisk-users] Voice mail "maxmessage " setting per mail box   Bruce McAlister
• [asterisk-users] Voice mail "maxmessage " setting per mail box   Bruce McAlister
• [asterisk-users] Voice mail "maxmessage " setting per mail box   Bruce McAlister
• [asterisk-users] Voice mail "maxmessage " setting per mail box   Bruce McAlister
• [asterisk-users] Voice mail "maxmessage " setting per mail box   Bruce McAlister
• [asterisk-users] Debug help   McCann, Brian
• [asterisk-users] Avaya 9640 Convert to SIP (slightly off topic)   Ron McCarthy
• [asterisk-users] Xorcom Experience   Eric Merkel
• [asterisk-users] cat /proc/zaptel/*   Russ Meyerriecks
• [asterisk-users] RPID on called party   Mark Michelson
• [asterisk-users] A matter of context   Eddie Mikell
• [asterisk-users] A matter of Context   Eddie Mikell
• [asterisk-users] I figured it out!!   Eddie Mikell
• [asterisk-users] Problems with Fax over TDM410P   Barry Miller
• [asterisk-users] Changing storm-prevention behaviour in logger.conf   Barry Miller
• [asterisk-users] Changing storm-prevention behaviour in logger.conf   Barry Miller
• [asterisk-users] asterisk-gplonly dependency in asterisk-addons RPM   Chris Miller
• [asterisk-users] asterisk-gplonly dependency in asterisk-addons RPM   Chris Miller
• [asterisk-users] Dynamic agent showing as "Invalid"   Miguel Molina
• [asterisk-users] Is svn.asterisk.org down ?   Miguel Molina
• [asterisk-users] Help with FastAGI server in Windows   Miguel Molina
• [asterisk-users] Gateway E1 <=> Asterisk ?   Luis Morales
• [asterisk-users] Being attacked by an Amazon EC2 ...   Steve Murphy
• [asterisk-users] Being attacked by an Amazon EC2 ...   Steve Murphy
• [asterisk-users] Asterisk choking on voice messages announcements   Steve Murphy
• [asterisk-users] Asterisk choking on voice messages announcements   Steve Murphy
• [asterisk-users] dial extension and play sound file from shell on asterisk server?   Brian J. Murrell
• [asterisk-users] two FreePBX servers with load balancing   Hector Muñoz
• [asterisk-users] kamailio   Hector Muñoz
• [asterisk-users] kamailio   Hector Muñoz
• [asterisk-users] mISDN installation via yum   Michael Nausch
• [asterisk-users] mISDN installation via yum   Michael Nausch
• [asterisk-users] mISDN installation via yum   Michael Nausch
• [asterisk-users] Strange Centos Problem with Dahdi installation   Tim Nelson
• [asterisk-users] jitterbuffer   Tim Nelson
• [asterisk-users] jitterbuffer   Tim Nelson
• [asterisk-users] jitterbuffer   Tim Nelson
• [asterisk-users] jitterbuffer   Tim Nelson
• [asterisk-users] cause 66 - Channel not implemented   Tim Nelson
• [asterisk-users] PRI Gurus ONLY - Too complex of an issue   Tim Nelson
• [asterisk-users] IAX connection slow between 1.4 and 1.6 dists   Tim Nelson
• [asterisk-users] IAX connection slow between 1.4 and 1.6 dists   Tim Nelson
• [asterisk-users] E3 Card on Asterisk ?   Tim Nelson
• [asterisk-users] Gateway E1 <=> Asterisk ?   Tim Nelson
• [asterisk-users] Gateway E1 <=> Asterisk ?   Tim Nelson
• [asterisk-users] SpiderMux?   Tim Nelson
• [asterisk-users] SpiderMux?   Tim Nelson
• [asterisk-users] B410P and DTMF   matthieu Nicaise
• [asterisk-users] Press release: Virtual Communication Clouds:: New feature in Asterisk 1.8   Danny Nicholas
• [asterisk-users] canary_thread   Danny Nicholas
• [asterisk-users] How set debug file for RxFax application   Danny Nicholas
• [asterisk-users] asterisk start with php   Danny Nicholas
• [asterisk-users] Continuous bothering message -- Remote UNIXconnection disconnected   Danny Nicholas
• [asterisk-users] External Extension with Extension   Danny Nicholas
• [asterisk-users] Limit Number Of Simultaneous Outbound SIP Calls   Danny Nicholas
• [asterisk-users] dialplan checker   Danny Nicholas
• [asterisk-users] Asterisk hangup all outging calls after32 seconds   Danny Nicholas
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   Danny Nicholas
• [asterisk-users] run script after completed   Danny Nicholas
• [asterisk-users] Being attacked by an Amazon EC2 ...   Danny Nicholas
• [asterisk-users] Monitoring calls via sound card   Danny Nicholas
• [asterisk-users] Time variables in system application   Danny Nicholas
• [asterisk-users] Do AMI Events have timestamps?   Danny Nicholas
• [asterisk-users] Time variables in system application   Danny Nicholas
• [asterisk-users] Possible AGI bug?   Danny Nicholas
• [asterisk-users] Time variables in system application   Danny Nicholas
• [asterisk-users] Do AMI Events have timestamps?   Danny Nicholas
• [asterisk-users] Time variables in system application   Danny Nicholas
• [asterisk-users] Vestec vs Lumenvox   Danny Nicholas
• [asterisk-users] IAX connection slow between 1.4 and 1.6 dists   Danny Nicholas
• [asterisk-users] Recording music in Queue   Danny Nicholas
• [asterisk-users] IAX connection slow between 1.4 and 1.6 dists   Danny Nicholas
• [asterisk-users] IAX connection slow between 1.4 and 1.6 dists   Danny Nicholas
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Danny Nicholas
• [asterisk-users] Odd Issue With Polycom Phones   Danny Nicholas
• [asterisk-users] Read Timeout   Danny Nicholas
• [asterisk-users] Calls drop after 20 seconds   Danny Nicholas
• [asterisk-users] Voice mail "maxmessage " setting per mail box   Danny Nicholas
• [asterisk-users] Voice mail "maxmessage " setting per mail box   Danny Nicholas
• [asterisk-users] Read Timeout   Danny Nicholas
• [asterisk-users] Voice mail "maxmessage " setting per mail box   Danny Nicholas
• [asterisk-users] 1.6.2 No "soft hangup"?   Danny Nicholas
• [asterisk-users] Swaping out phones.   Danny Nicholas
• [asterisk-users] Hangup after n seconds using originate ?   Danny Nicholas
• [asterisk-users] More efficient dial plan for a list of selectiveinbound numbers   Danny Nicholas
• [asterisk-users] Follow-me to my answering machine :-(   Danny Nicholas
• [asterisk-users] Follow-me to my answering machine :-(   Danny Nicholas
• [asterisk-users] Playback all the sound files   Danny Nicholas
• [asterisk-users] Inbound route question   Danny Nicholas
• [asterisk-users] Inbound route question   Danny Nicholas
• [asterisk-users] Building Asterisk-RPM for 1.4.24.1   Danny Nicholas
• [asterisk-users] Connect 2 asterisks servers   Danny Nicholas
• [asterisk-users] Detect if a Number is up or not   Danny Nicholas
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Danny Nicholas
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Danny Nicholas
• [asterisk-users] command-line dialplan "compiler"   Danny Nicholas
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Danny Nicholas
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Danny Nicholas
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Danny Nicholas
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Danny Nicholas
• [asterisk-users] simple dialplan question   Danny Nicholas
• [asterisk-users] Strange Error -- ASterisk 1.6   Danny Nicholas
• [asterisk-users] Issue with (pattern) matching extension   Danny Nicholas
• [asterisk-users] Dropping incompatible voice frame   Danny Nicholas
• [asterisk-users] Issue with (pattern) matching extension   Danny Nicholas
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Danny Nicholas
• [asterisk-users] Code in extensions.conf to leave a voice mail inanother PBX ?!   Danny Nicholas
• [asterisk-users] Issue with (pattern) matching extension   Danny Nicholas
• [asterisk-users] Issue with (pattern) matching extension   Danny Nicholas
• [asterisk-users] scratchy sound   Oliver Nittka
• [asterisk-users] SIP Connection Question   Dr. Kenneth Noisewater
• [asterisk-users] asterisk start with php   Dr. Kenneth Noisewater
• [asterisk-users] SIP Connection Question   Kenneth Noisewater
• [asterisk-users] External Extension with Extension   Kenneth Noisewater
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   John Novack
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   John Novack
• [asterisk-users] How to set up Fax on Asterisk - Using analog Fax machines and HT502 (or FXS of a Digium TDM410P)   John Novack
• [asterisk-users] E3 Card on Asterisk ?   John Novack
• [asterisk-users] Inbound route question   Alejandro Cabrera Obed
• [asterisk-users] Inbound route question   Alejandro Cabrera Obed
• [asterisk-users] Gosub replacement within AEL2 dialplans   Olivier
• [asterisk-users] Gosub replacement within AEL2 dialplans   Olivier
• [asterisk-users] Which rule for Asterisk to Asterisk-addons compatibility ?   Olivier
• [asterisk-users] OT - S450ip and R-key transfer   Olivier
• [asterisk-users] Change in menuselect handling of sound files (in 1.6.1.X)   Olivier
• [asterisk-users] Dahdi, junghanns and qozap   Olivier
• [asterisk-users] Dahdi, junghanns and qozap   Olivier
• [asterisk-users] Dahdi, junghanns and qozap   Olivier
• [asterisk-users] dahdi_scan and OctoBRI. Bug or feature ?   Olivier
• [asterisk-users] Is svn.asterisk.org down ?   Olivier
• [asterisk-users] Using chan_lcr (and mISDN v2) ?   Olivier
• [asterisk-users] Bug or feature: cdr_odbc.conf.sample   Olivier
• [asterisk-users] Asterisk-stat - Bugs   Olivier
• [asterisk-users] The best way to stop an ongoing call   Olivier
• [asterisk-users] How to tell if a channel is on hold or not from diaplan ?   Olivier
• [asterisk-users] Feature Request - SoftHangup with delayed playback option   Olivier
• [asterisk-users] 1.6.1.18 : app_voicemail is calling sendmail without any argument   Olivier
• [asterisk-users] 1.6.2 - Pickup and SIP Replaces header   Olivier
• [asterisk-users] 1.6.2 - Pickup and SIP Replaces header   Olivier
• [asterisk-users] How to disable dialog-info based call pickups (Was: Re: 1.6.2 - Pickup and SIP Replaces header)   Olivier
• [asterisk-users] Message notification without MWI   Olivier
• [asterisk-users] Asterisk and Patton   Olivier
• [asterisk-users] SIP Dialplan Failover Solution   Alexandru Oniciuc
• [asterisk-users] Extensions Reload | Asterisk Freezes ? 1.4   Positively Optimistic
• [asterisk-users] Extensions Reload | Asterisk Freezes ? 1.4   Positively Optimistic
• [asterisk-users] Being attacked by an Amazon EC2 ...   Chris Owen
• [asterisk-users] Being attacked by an Amazon EC2 ...   Chris Owen
• [asterisk-users] SIP Outdial Not Detecting Ringing Line   Deric Page
• [asterisk-users] Limit Number Of Simultaneous Outbound SIP Calls   Deric Page
• [asterisk-users] Limit Number Of Simultaneous Outbound SIP Calls   Deric Page
• [asterisk-users] AMD reporting NOTSURE most of the time   Baji Panchumarti
• [asterisk-users] Regarding remote registration of SIP user on zoiper   Vinod Parameswaran
• [asterisk-users] asterisk 1.6.1/1.6.2 binary packages   Jason Parker
• [asterisk-users] asterisk 1.6.1/1.6.2 binary packages   Jason Parker
• [asterisk-users] mISDN installation via yum   Jason Parker
• [asterisk-users] Change in menuselect handling of sound files (in 1.6.1.X)   Jason Parker
• [asterisk-users] t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED   Darshaka Pathirana
• [asterisk-users] t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED   Darshaka Pathirana
• [asterisk-users] Asterisk in Debian/Lenny without Junghanns.net support?   Darshaka Pathirana
• [asterisk-users] SNOM M9 base station A to base station B   Justin Paton
• [asterisk-users] Calls drop after 20 seconds   Peder
• [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1   Peder
• [asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!   Peder
• [asterisk-users] meetme / upgrade to 1.6.2.6   Thomas Perron
• [asterisk-users] Dropped Calls   Peter
• [asterisk-users] GXW4024   Peter
• [asterisk-users] Fwd: Re: SpiderMux?   Peter
• [asterisk-users] Subscribe to a MWI when acting as a SIP client?   Alfredo Peña
• [asterisk-users] Being attacked by an Amazon EC2 ...   Fred Posner
• [asterisk-users] Being attacked by an Amazon EC2 ...   Fred Posner
• [asterisk-users] Being attacked by an Amazon EC2 ...   Fred Posner
• [asterisk-users] Being attacked by an Amazon EC2 ...   Fred Posner
• [asterisk-users] Flood of REGISTERs - attack?   Fred Posner
• [asterisk-users] Being attacked by an Amazon EC2 ...   Fred Posner
• [asterisk-users] Being attacked by an Amazon EC2 ...   Fred Posner
• [asterisk-users] Amazon EC2 SIP floods - you can help   Fred Posner
• [asterisk-users] Amazon EC2 SIP floods - you can help   Fred Posner
• [asterisk-users] Being attacked by an Amazon EC2 ...   Fred Posner
• [asterisk-users] Being attacked by an Amazon EC2 ...   Fred Posner
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Fred Posner
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Fred Posner
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Fred Posner
• [asterisk-users] Issue with (pattern) matching extension   Philip Prindeville
• [asterisk-users] Issue with (pattern) matching extension   Philip Prindeville
• [asterisk-users] Taqua users out there?   Philip A. Prindeville
• [asterisk-users] Issue with (pattern) matching extension   Philip A. Prindeville
• [asterisk-users] Issue with (pattern) matching extension   Philip A. Prindeville
• [asterisk-users] Issue with (pattern) matching extension   Philip A. Prindeville
• [asterisk-users] Issue with (pattern) matching extension   Philip A. Prindeville
• [asterisk-users] problem compiling asterisk with cdr_odbc   Nathan Pryor
• [asterisk-users] problem compiling asterisk with cdr_odbc   Nathan Pryor
• [asterisk-users] problem compiling asterisk with cdr_odbc   Nathan Pryor
• [asterisk-users] moh files not playing in sort order   Nathan Pryor
• [asterisk-users] asterisk core dumps after cdr database writes using odbc   Nathan Pryor
• [asterisk-users] Asterisk and spandsp fax problem   Fulajtár Pál
• [asterisk-users] Problem with Callfiles   Edwin Quijada
• [asterisk-users] FastAGiin Windows Server   Edwin Quijada
• [asterisk-users] FastAGiin Windows Server   Edwin Quijada
• [asterisk-users] FastAGiin Windows Server   Edwin Quijada
• [asterisk-users] FastAGiin Windows Server   Edwin Quijada
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Edwin Quijada
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Edwin Quijada
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Edwin Quijada
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Edwin Quijada
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Edwin Quijada
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Edwin Quijada
• [asterisk-users] Help with FastAGI server in Windows   Edwin Quijada
• [asterisk-users] Being attacked by an Amazon EC2 ...   David Quinton
• [asterisk-users] Being attacked by an Amazon EC2 ...   David Quinton
• [asterisk-users] Asterisk a-law header missing?   Pham Quy
• [asterisk-users] How to log into separate file   Pham Quy
• [asterisk-users] How to log into separate file   Pham Quy
• [asterisk-users] asterisk segmentation fault   Pham Quy
• [asterisk-users] Does 'file' command work with asterisk genereted alaw file   Pham Quy
• [asterisk-users] Does 'file' command work with asterisk genereted alaw file   Pham Quy
• [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8   Randy R
• [asterisk-users] Anyone coming to Paris next week for AstriEurope?   Randy R
• [asterisk-users] Anyone coming to Paris next week for AstriEurope?   Randy R
• [asterisk-users] Anyone coming to Paris next week for AstriEurope?   Randy R
• [asterisk-users] Friday Apr 9th 2010 @ 12 Noon EDT: More Cloud Telephony   Randy R
• [asterisk-users] Being attacked by an Amazon EC2 ...   Randy R
• [asterisk-users] Being attacked by an Amazon EC2 ...   Randy R
• [asterisk-users] Being attacked by an Amazon EC2   Randy R
• [asterisk-users] Being attacked by an Amazon EC2 ...   Randy R
• [asterisk-users] Being attacked by an Amazon EC2 ...   Randy R
• [asterisk-users] Friday April 16 @12 Noon EDT - Tim's Excellent Island Telephony Adventure, AstriEurop, and more EC2 rant   Randy R
• [asterisk-users] Amazon EC2 SIP floods - you can help   Randy R
• [asterisk-users] Amazon EC2 SIP floods - you can help   Randy R
• [asterisk-users] Amazon EC2 SIP floods - you can help   Randy R
• [asterisk-users] Amazon EC2 SIP floods - you can help   Randy R
• [asterisk-users] Amazon EC2 SIP floods - you can help   Randy R
• [asterisk-users] Amazon EC2 SIP floods - you can help   Randy R
• [asterisk-users] Evaluating Asterisk   Randy R
• [asterisk-users] Being attacked by an Amazon EC2 ...   Randy R
• [asterisk-users] Being attacked by an Amazon EC2 ...   Randy R
• [asterisk-users] Being attacked by an Amazon EC2 ...   Randy R
• [asterisk-users] VUC Friday: Bill Miller, former VP of Product Management   Randy R
• [asterisk-users] Friday 12 Noon EDT: Media5fone Mobile SIP Client Symbian S60 & iPhone   Randy R
• [asterisk-users] Problem with Sangoma A104 and euroisdn pri   RESEARCH
• [asterisk-users] PSTN issues   Balu Raman
• [asterisk-users] Calls drop after 20 seconds   Alejandro Recarey
• [asterisk-users] Calls drop after 20 seconds   Alejandro Recarey
• [asterisk-users] Unable to load cdr_adaptive_odbc.so   Alejandro Recarey
• [asterisk-users] Unable to load cdr_adaptive_odbc.so   Alejandro Recarey
• [asterisk-users] Time difference in CSV CDR's and MySQL CDR's   Alejandro Recarey
• [asterisk-users] Calls drop after 20 seconds   Alejandro Recarey
• [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension   Alejandro Recarey
• [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension   Alejandro Recarey
• [asterisk-users] Asterisk not recognizing ACK from an OK message? Help debuging SIP retransmit problem   Alejandro Recarey
• [asterisk-users] Asterisk not recognizing ACK from an OK message? Help debuging SIP retransmit problem   Alejandro Recarey
• [asterisk-users] Asterisk not recognizing ACK from an OK message?Help debuging SIP retransmit problem   Alejandro Recarey
• [asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution   JR Richardson
• [asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution   JR Richardson
• [asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution   JR Richardson
• [asterisk-users] Being attacked by an Amazon EC2   JR Richardson
• [asterisk-users] [PATCH] Make Queue announcements more consistent (1.4.26.2)   Matt Riddell
• [asterisk-users] Follow-me to my answering machine :-(   Matt Riddell
• [asterisk-users] RTP over TCP   Matt Riddell
• [asterisk-users] Installing For AsteirskAddon   Matt Riddell
• [asterisk-users] Follow-me to my answering machine :-(   Matt Riddell
• [asterisk-users] misdn accountcode?   Alexandre Rodrigues
• [asterisk-users] RPID on called party   "Juan E. Rodríguez"
• [asterisk-users] How set debug file for RxFax application   "Juan E. Rodríguez"
• [asterisk-users] SIP Connection Question   Juan E. Rodríguez
• [asterisk-users] SIP Connection Question   Juan E. Rodríguez
• [asterisk-users] 1.6.0 verses 1.6.2   John Rose
• [asterisk-users] tones detection   John Rose
• [asterisk-users] 1.6.0 verses 1.6.2   John Rose
• [asterisk-users] Redone setup, bizare problems   Nicolas Ross
• [asterisk-users] zapg723toslin did not update samples   Shaun Ruffell
• [asterisk-users] B400P and A1200P changes card order   Shaun Ruffell
• [asterisk-users] Asterisk choking on voice messages announcements   Shaun Ruffell
• [asterisk-users] asterisk 1.6.1/1.6.2 binary packages   Pablo Ruiz
• [asterisk-users] asterisk 1.6.1/1.6.2 binary packages   Pablo Ruiz
• [asterisk-users] asterisk 1.6.1/1.6.2 binary packages   Pablo Ruiz
• [asterisk-users] Strange Centos Problem with Dahdi installation   ABBAS SHAKEEL
• [asterisk-users] Strange Centos Problem with Dahdi installation   ABBAS SHAKEEL
• [asterisk-users] Strange Centos Problem with Dahdi installation   ABBAS SHAKEEL
• [asterisk-users] Strange Centos Problem with Dahdi installation   ABBAS SHAKEEL
• [asterisk-users] Detect if a Number is up or not   ABBAS SHAKEEL
• [asterisk-users] Detect if a Number is up or not   ABBAS SHAKEEL
• [asterisk-users] Detect if a Number is up or not   ABBAS SHAKEEL
• [asterisk-users] Detect if a Number is up or not   ABBAS SHAKEEL
• [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286   Raimund Sacherer
• [asterisk-users] G.729 Codec problem.   Arun Sasidhar
• [asterisk-users] Sending RTP media to a different server than SIP Signaling   Tarek Sawah
• [asterisk-users] Sending RTP media to a different server than SIP Signaling   Tarek Sawah
• [asterisk-users] Sending RTP media to a different server than SIP Signaling   Tarek Sawah
• [asterisk-users] Inbound route question   Tarek Sawah
• [asterisk-users] Strange Invite issue   Tarek Sawah
• [asterisk-users] Calls Dropping   Tarek Sawah
• [asterisk-users] Strange Invite issue   Tarek Sawah
• [asterisk-users] Strange Invite issue   Tarek Sawah
• [asterisk-users] Asterisk hangup all outging calls after 32 seconds   Gregory Miles Blumenthal Scharf
• [asterisk-users] canary_thread   Stefan Schmidt
• [asterisk-users] scratchy sound   Stefan Schmidt
• [asterisk-users] Calls drop after 20 seconds   Stefan Schmidt
• [asterisk-users] Calls drop after 20 seconds   Stefan Schmidt
• [asterisk-users] VoIP monitoring tools   Stefan Schmidt
• [asterisk-users] Strange Centos Problem with Dahdi installation   Warren Selby
• [asterisk-users] OT: Wireless headset / phone combination   Warren Selby
• [asterisk-users] Converting GSM calls to SIP   Warren Selby
• [asterisk-users] Odd Issue With Polycom Phones   Warren Selby
• [asterisk-users] Dial plan question.   Warren Selby
• [asterisk-users] Initial audio dropping   David Shauger
• [asterisk-users] Embedded IAX   Bill Shaw
• [asterisk-users] Being attacked by an Amazon EC2 ...   Stuart Sheldon
• [asterisk-users] Amazon EC2 SIP floods - you can help   Stuart Sheldon
• [asterisk-users] Amazon EC2 SIP floods - you can help   Stuart Sheldon
• [asterisk-users] Being attacked by an Amazon EC2 ...   Stuart Sheldon
• [asterisk-users] IVR menu sound processing for AMR and GSM + live test available   Arkadi Shishlov
• [asterisk-users] IVR menu sound processing for AMR and GSM + live test available   Arkadi Shishlov
• [asterisk-users] run script after completed   Arkadi Shishlov
• [asterisk-users] Set CDR amaflags not work   Zhang Shukun
• [asterisk-users] Detect if a Number is up or not   Zhang Shukun
• [asterisk-users] G729 exhaustion conditions   Zhang Shukun
• [asterisk-users] Time difference in CSV CDR's and MySQL CDR's   Zhang Shukun
• [asterisk-users] Time difference in CSV CDR's and MySQL CDR's   Zhang Shukun
• [asterisk-users] Embedded IAX   Moises Silva
• [asterisk-users] Opportunity to author Asterisk books- Packt Publishing.   Kshipra Singh
• [asterisk-users] Interesting One Way Audio   Prince Singh
• [asterisk-users] BN8S0, dahdi, wcb4xxp   Claire Sinn
• [asterisk-users] BN8S0, dahdi, wcb4xxp   Claire Sinn
• [asterisk-users] BN8S0, dahdi, wcb4xxp   Claire Sinn
• [asterisk-users] BN8S0, dahdi, wcb4xxp   Claire Sinn
• [asterisk-users] SIP Dialplan Failover Solution   Aurimas Skirgaila
• [asterisk-users] is it possible to connect Digium TE420 and Cisco card?   Aurimas Skirgaila
• [asterisk-users] Manipulating audio in asterisk   Slawek Sloma
• [asterisk-users] problem compiling asterisk with cdr_odbc   Jared Smith
• [asterisk-users] Do AMI Events have timestamps?   Jared Smith
• [asterisk-users] Transfer_CONTEXT behaviour   Jared Smith
• [asterisk-users] Asterisk/Polycom Dialed Party Name   Jared Smith
• [asterisk-users] Transfer_CONTEXT behaviour   Jared Smith
• [asterisk-users] DIALSTATUS variable and qualify=no   Jared Smith
• [asterisk-users] Voice mail "maxmessage " setting per mail box   Jared Smith
• [asterisk-users] 1.6.2 No "soft hangup"?   Jared Smith
• [asterisk-users] Portech MV-374 does not register behind NAT   Jared Smith
• [asterisk-users] Asterisk/Polycom Dialed Party Name   Marc Smith
• [asterisk-users] Being attacked by an Amazon EC2 ...   Mark Smith
• [asterisk-users] Being attacked by an Amazon EC2 ...   Mark Smith
• [asterisk-users] MeetMe Options with S(10)L(100)   Chandrakant Solanki
• [asterisk-users] [Conference] Audio/Video   Jamie A. Stapleton
• [asterisk-users] Problems for Skype for Asterisk   Jamie A. Stapleton
• [asterisk-users] Continuing after a TIMEOUT(absolute)   Brendan Sterne
• [asterisk-users] Continuing after a TIMEOUT(absolute)   Brendan Sterne
• [asterisk-users] Being attacked by an Amazon EC2 ...   Tom Stordy-Allison
• [asterisk-users] Being attacked by an Amazon EC2 ...   Tom Stordy-Allison
• [asterisk-users] Being attacked by an Amazon EC2 ...   Tom Stordy-Allison
• [asterisk-users] Flood of REGISTERs - attack?   Tom Stordy-Allison
• [asterisk-users] Asterisk a-law header missing?   Quy Pham Sy
• [asterisk-users] How to log into separate file   Quy Pham Sy
• [asterisk-users] Converting GSM calls to SIP   Tonty T
• [asterisk-users] Converting GSM calls to SIP   Tonty T
• [asterisk-users] Converting GSM calls to SIP   Tonty T
• [asterisk-users] meetme / upgrade to 1.6.2.6   Tonty T
• [asterisk-users] Being attacked by an Amazon EC2 ...   Administrator TOOTAI
• [asterisk-users] DAHDI-Linux and DAHDI-Tools 2.3.0 Released   Asterisk Development Team
• [asterisk-users] Asterisk-Addons 1.4.11, 1.6.0.5, 1.6.1.3, and 1.6.2.1 Now Available   Asterisk Development Team
• [asterisk-users] SIP equivalent of zap "c" option   Mark G. Thomas
• [asterisk-users] Asterisk + DRBD Performance   Jonathan Thurman
• [asterisk-users] iptables miss up phone calls if not used properly   Jonathan Thurman
• [asterisk-users] Realtime changes not reflected realtime   Jonathan Thurman
• [asterisk-users] Realtime changes not reflected realtime   Jonathan Thurman
• [asterisk-users] Realtime changes not reflected realtime   Jonathan Thurman
• [asterisk-users] High Availability - Shared Database - Ideas?   Jonathan Thurman
• [asterisk-users] Gateway E1 <=> Asterisk ?   Jonathan Thurman
• [asterisk-users] GXW4024   Jonathan Thurman
• [asterisk-users] Reset personal voicemail settings   Felix Tiefenthaler
• [asterisk-users] Question about MaxRetries in the Asterisk Outgoing folder   John Timms
• [asterisk-users] dialplan question   Vasiliy G Tolstov
• [asterisk-users] dialplan question   Vasiliy G Tolstov
• [asterisk-users] mysql realtime schema   Vasiliy G Tolstov
• [asterisk-users] D-Channel Span Up without Down   Steve Totaro
• [asterisk-users] PSTN issues   Steve Totaro
• [asterisk-users] QUICKLY - What is the command to change .wmv to Asterisk compatible format using ffmpeg? or any other tool?   Steve Totaro
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Steve Totaro
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Steve Totaro
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Steve Totaro
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Steve Totaro
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Steve Totaro
• [asterisk-users] AGI, FASTAGI or Windows Voice Server   Steve Totaro
• [asterisk-users] Security tests   Steve Totaro
• [asterisk-users] Amazon EC2 SIP floods - you can help   Rob Townley
• [asterisk-users] dial extension and play sound file from shell on asterisk server?   Duncan Turnbull
• [asterisk-users] E3 Card on Asterisk ?   Steve Underwood
• [asterisk-users] Odd Issue With Polycom Phones]   Gord Urquhart
• [asterisk-users] RPID on called party   Ondrej Valousek
• [asterisk-users] Follow-me to my answering machine :-(   Vince Vielhaber
• [asterisk-users] scratchy sound   Vieri
• [asterisk-users] cause 66 - Channel not implemented   Vieri
• [asterisk-users] SIP one-way audio   Vieri
• [asterisk-users] Calls drop after 20 seconds   Vieri
• [asterisk-users] hardware clock drift and CDR   Vieri
• [asterisk-users] SIP gain   Vieri
• [asterisk-users] hardware clock drift and CDR   Vieri
• [asterisk-users] hardware clock drift and CDR   Vieri
• [asterisk-users] simple dialplan question   Vieri
• [asterisk-users] sip jitter buffer   Vieri
• [asterisk-users] Dropping incompatible voice frame   Vieri
• [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286   Vieri
• [asterisk-users] IAX trunks and audio codecs   Vieri
• [asterisk-users] canary_thread   Marcus Vinicius
• [asterisk-users] Asterisk and MWI with Exchange 2010   Jay Vocaire
• [asterisk-users] Odd Issue With Polycom Phones   Jay Vocaire
• [asterisk-users] Odd Issue With Polycom Phones   Jay Vocaire
• [asterisk-users] Odd Issue With Polycom Phones   Jay Vocaire
• [asterisk-users] Odd Issue With Polycom Phones   Jay Vocaire
• [asterisk-users] Extensions Reload | Asterisk Freezes ? 1.4   Kristijan Vrban
• [asterisk-users] Asterisk stopping for no reason   Alexandre Vézina
• [asterisk-users] Asterisk stopping for no reason   Alexandre Vézina
• [asterisk-users] D-Channel Span Up without Down   Jason Walker
• [asterisk-users] Recording music in Queue   Jason Walker
• [asterisk-users] Continuous bothering message -- Remote UNIXconnection disconnected   Watkins, Bradley
• [asterisk-users] Interesting One Way Audio   Thermal Wetland
• [asterisk-users] Interesting One Way Audio   Thermal Wetland
• [asterisk-users] Put a call on hold with Manager   Thermal Wetland
• [asterisk-users] Asterisk not recognizing ACK from an OK message?Help debuging SIP retransmit problem   David White
• [asterisk-users] Connect 2 asterisks servers   David White
• [asterisk-users] Strange Invite issue   David White
• [asterisk-users] Strange Invite issue   David White
• [asterisk-users] over running my did's   Darren Wiebe
• [asterisk-users] Rebooting Polycom's - Could not create address for 'XXXX'   Jake Wilson
• [asterisk-users] Multiple Parking Lots in Asterisk 1.6.x   Michael Wilson
• [asterisk-users] VoIP monitoring tools   Michael Wilson
• [asterisk-users] Evaluating Asterisk   Stephen Wingfield
• [asterisk-users] Asterisk script to repeat dial of a number   Shaun Wingrin
• [asterisk-users] Problem with Sangoma A104 and euroisdn pri   Jaap Winius
• [asterisk-users] Problem with Sangoma A104 and euroisdn pri   Jaap Winius
• [asterisk-users] Problem with Sangoma A104 and euroisdn pri - SOLVED   Jaap Winius
• [asterisk-users] cat /proc/zaptel/*   Jaap Winius
• [asterisk-users] cat /proc/zaptel/*   Jaap Winius
• [asterisk-users] cat /proc/zaptel/*   Jaap Winius
• [asterisk-users] Being attacked by an Amazon EC2 ...   Hans Witvliet
• [asterisk-users] Being attacked by an Amazon EC2 ...   Hans Witvliet
• [asterisk-users] RTP over TCP   Hans Witvliet
• [asterisk-users] Slightly OT: OMA DM Solution   Jay R. Worthington
• [asterisk-users] Slightly OT: OMA DM Solution   Jay R. Worthington
• [asterisk-users] Converting GSM calls to SIP   Vahan Yerkanian
• [asterisk-users] SS7 over an FXO interface   Vahan Yerkanian
• [asterisk-users] Continuous bothering message -- Remote UNIX connection disconnected   Zeeshan Zakaria
• [asterisk-users] Continuous bothering message -- Remote UNIX connection disconnected   Zeeshan Zakaria
• [asterisk-users] Continuous bothering message -- Remote UNIX connection disconnected   Zeeshan Zakaria
• [asterisk-users] Debug help   Zeeshan Zakaria
• [asterisk-users] AGI + Dial + stream file ?   Zeeshan Zakaria
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   Zeeshan Zakaria
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   Zeeshan Zakaria
• [asterisk-users] Being attacked by an Amazon EC2 ...   Zeeshan Zakaria
• [asterisk-users] Being attacked by an Amazon EC2 ...   Zeeshan Zakaria
• [asterisk-users] Being attacked by an Amazon EC2 ...   Zeeshan Zakaria
• [asterisk-users] Being attacked by an Amazon EC2 ...   Zeeshan Zakaria
• [asterisk-users] Being attacked by an Amazon EC2 ...   Zeeshan Zakaria
• [asterisk-users] Being attacked by an Amazon EC2 ...   Zeeshan Zakaria
• [asterisk-users] All incoming calls landing in [customers] context   Zeeshan Zakaria
• [asterisk-users] Asterisk DIES with no trace. PLEASE   Zeeshan Zakaria
• [asterisk-users] Asterisk DIES with no trace. PLEASE   Zeeshan Zakaria
• [asterisk-users] Conference Meetme   Zeeshan Zakaria
• [asterisk-users] Delay the HungUp   Zeeshan Zakaria
• [asterisk-users] How to do analog e&m on asterisk?   Zeeshan Zakaria
• [asterisk-users] How to do analog e&m on asterisk?   Zeeshan Zakaria
• [asterisk-users] How to do analog e&m on asterisk?   Zeeshan Zakaria
• [asterisk-users] RTP over TCP   Zeeshan Zakaria
• [asterisk-users] RTP over TCP   Zeeshan Zakaria
• [asterisk-users] RTP over TCP   Zeeshan Zakaria
• [asterisk-users] RTP over TCP   Zeeshan Zakaria
• [asterisk-users] Being attacked by an Amazon EC2 ...   Norbert Zawodsky
• [asterisk-users] Being attacked by an Amazon EC2 ...   Norbert Zawodsky
• [asterisk-users] Being attacked by an Amazon EC2 ...   Norbert Zawodsky
• [asterisk-users] AGI <==> DeadAGI   Redouane Zerargui
• [asterisk-users] AGI <==> DeadAGI   Redouane Zerargui
• [asterisk-users] asterisk-users Digest, Vol 69, Issue 16   Alan Zheng
• [asterisk-users] Fwd: Re: SpiderMux?   Zoa
• [asterisk-users] Avaya UUI   Zsotya
• [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8   --[ UxBoD ]--
• [asterisk-users] [OT] phpagi help   --[ UxBoD ]--
• [asterisk-users] [OT] phpagi help   --[ UxBoD ]--
• [asterisk-users] Being attacked by an Amazon EC2 ...   --[ UxBoD ]--
• [asterisk-users] Being attacked by an Amazon EC2 ...   --[ UxBoD ]--
• [asterisk-users] Being attacked by an Amazon EC2 ...   --[ UxBoD ]--
• [asterisk-users] Being attacked by an Amazon EC2 ...   --[ UxBoD ]--
• [asterisk-users] Being attacked by an Amazon EC2 ...   --[ UxBoD ]--
• [asterisk-users] Being attacked by an Amazon EC2 ...   --[ UxBoD ]--
• [asterisk-users] Being attacked by an Amazon EC2 ...   --[ UxBoD ]--
• [asterisk-users] SS7 over an FXO interface   mosbah abdelkader
• [asterisk-users] SS7 over an FXO interface   mosbah abdelkader
• [asterisk-users] softphone help   ayodele abejide
• [asterisk-users] How can I record the conversations in a conference call?   Renato bianchini
• [asterisk-users] On CLI SIP don't appear   Renato bianchini
• [asterisk-users] problem of "when memory become 50% or more then sound become noisy?"   kamrun nahar bina
• [asterisk-users] problem of "when memory become 50% or more then sound become noisy?"   kamrun nahar bina
• [asterisk-users] Avaya 9640 Convert to SIP (slightly off topic)   robert boardman
• [asterisk-users] PrivacyManager   robert boardman
• [asterisk-users] asterisk 1.6.1/1.6.2 binary packages   bruce bruce
• [asterisk-users] asterisk 1.6.1/1.6.2 binary packages   bruce bruce
• [asterisk-users] asterisk 1.6.1/1.6.2 binary packages   bruce bruce
• [asterisk-users] call files in 1.6   bruce bruce
• [asterisk-users] Access denied for user 'a2billinguser   bruce bruce
• [asterisk-users] D-Channel Span Up without Down   bruce bruce
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   bruce bruce
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   bruce bruce
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   bruce bruce
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   bruce bruce
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   bruce bruce
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   bruce bruce
• [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls   bruce bruce
• [asterisk-users] PRI - Native ZAP bridge fails - Is this my patch?   bruce bruce
• [asterisk-users] Sending RTP media to a different server than SIP Signaling   bruce bruce
• [asterisk-users] Asterisk script to repeat dial of a number   bruce bruce
• [asterisk-users] Sending RTP media to a different server than SIP Signaling   bruce bruce
• [asterisk-users] Sending RTP media to a different server than SIP Signaling   bruce bruce
• [asterisk-users] Sending RTP media to a different server than SIP Signaling   bruce bruce
• [asterisk-users] PRI - Native ZAP bridge fails - Is this my patch?   bruce bruce
• [asterisk-users] PRI Gurus ONLY - Too complex of an issue   bruce bruce
• [asterisk-users] PRI Gurus ONLY - Too complex of an issue   bruce bruce
• [asterisk-users] PRI Gurus ONLY - Too complex of an issue   bruce bruce
• [asterisk-users] PRI Gurus ONLY - Too complex of an issue   bruce bruce
• [asterisk-users] PRI TBCT - Practical Experience, Anybody?   bruce bruce
• [asterisk-users] PRI Gurus ONLY - Too complex of an issue   bruce bruce
• [asterisk-users] PRI Gurus ONLY - Too complex of an issue - SOLVED   bruce bruce
• [asterisk-users] PRI TBCT - Practical Experience, Anybody?   bruce bruce
• [asterisk-users] PRI Gurus ONLY - Too complex of an issue   bruce bruce
• [asterisk-users] Is restart of span a concern on PRI?   bruce bruce
• [asterisk-users] Being attacked by an Amazon EC2 ...   bruce bruce
• [asterisk-users] Is restart of span a concern on PRI?   bruce bruce
• [asterisk-users] Being attacked by an Amazon EC2 ...   bruce bruce
• [asterisk-users] X-lite direct sip call - Is it possible?   bruce bruce
• [asterisk-users] QUICKLY - What is the command to change .wmv to Asterisk compatible format using ffmpeg? or any other tool?   bruce bruce
• [asterisk-users] QUICKLY - What is the command to change .wmv to Asterisk compatible format using ffmpeg? or any other tool?   bruce bruce
• [asterisk-users] X-lite direct sip call - Is it possible?   bruce bruce
• [asterisk-users] Zap PRI failed with Cause 34 - Where to check for problems?   bruce bruce
• [asterisk-users] X-lite direct sip call - Is it possible?   bruce bruce
• [asterisk-users] Zap PRI failed with Cause 34 - Where to check for problems?   bruce bruce
• [asterisk-users] Portech MV-374 does not register   bruce bruce
• [asterisk-users] Portech MV-374 does not register   bruce bruce
• [asterisk-users] Asterisk choking on voice messages announcements   bruce bruce
• [asterisk-users] Asterisk choking on voice messages announcements   bruce bruce
• [asterisk-users] Asterisk choking on voice messages announcements   bruce bruce
• [asterisk-users] Asterisk choking on voice messages announcements   bruce bruce
• [asterisk-users] Asterisk choking on voice messages announcements   bruce bruce
• [asterisk-users] Asterisk choking on voice messages announcements   bruce bruce
• [asterisk-users] Asterisk choking on voice messages announcements   bruce bruce
• [asterisk-users] Asterisk choking on voice messages announcements   bruce bruce
• [asterisk-users] Portech MV-374 does not register behind NAT   bruce bruce
• [asterisk-users] Portech MV-374 does not register behind NAT   bruce bruce
• [asterisk-users] More efficient dial plan for a list of selective inbound numbers   bruce bruce
• [asterisk-users] RTP over TCP   bruce bruce
• [asterisk-users] How to debug the problem of Asterisk using so much of CPU percentage...?   bruce bruce
• [asterisk-users] PRI - Native ZAP bridge fails - Is this my patch?   bruce bruce
• [asterisk-users] Delay the HungUp   cbulist
• [asterisk-users] Delay the HungUp   cbulist
• [asterisk-users] Connect 2 asterisks servers   matheus coppetti
• [asterisk-users] CONNECTEDLINE(), progressinband=no and 183 before 180 (with latest trunk)   crjw
• [asterisk-users] Asterisk and spandsp fax problem   sean darcy
• [asterisk-users] 1.6.2.6: can't upgrade from 1.6.1.18   sean darcy
• [asterisk-users] 1.6.2 No "soft hangup"?   sean darcy
• [asterisk-users] Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?]   sean darcy
• [asterisk-users] Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?]   sean darcy
• [asterisk-users] dialplan   wassim darwich
• [asterisk-users] dialplan   wassim darwich
• [asterisk-users] jitterbuffer   dotnetdub
• [asterisk-users] jitterbuffer   dotnetdub
• [asterisk-users] jitterbuffer   dotnetdub
• [asterisk-users] asterisk start with php   salaheddine elharit
• [asterisk-users] asterisk start with php   salaheddine elharit
• [asterisk-users] asterisk start with php   salaheddine elharit
• [asterisk-users] ATA status intermittent   marcelo ferreira
• [asterisk-users] IAX Problem   bob gailer
• [asterisk-users] IAX Problem   bob gailer
• [asterisk-users] IAX Call Rejected (was IAX Problem)   bob gailer
• [asterisk-users] Asterisk load balancing and failover   huu giang
• [asterisk-users] Asterisk load balancing and failover   huu giang
• [asterisk-users] Cache sound files for faster processing   huu giang
• [asterisk-users] RES: Cache sound files for faster processing   huu giang
• [asterisk-users] Cache sound files for faster processing   huu giang
• [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)   huu giang
• [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)   huu giang
• [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)   huu giang
• [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)   huu giang
• [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)   huu giang
• [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)   huu giang
• [asterisk-users] VOIP at BerkeleyTIP-Global meeting on Sunday April 18 12N-3P, & April 27   giovanni_re
• [asterisk-users] Hangup after 1 second of ringing ?   mancyborg at gmail.com
• [asterisk-users] Hangup after n seconds using originate ?   mancyborg at gmail.com
• [asterisk-users] Hangup after n seconds using originate ?   mancyborg at gmail.com
• [asterisk-users] Hangup after n seconds using originate ?   mancyborg at gmail.com
• [asterisk-users] Conference Meetme   torintino1 at hotmail.com
• [asterisk-users] Conference Meetme   torintino1 at hotmail.com
• [asterisk-users] Asterisk and Archlinux   ik
• [asterisk-users] [asterisk users] asterisk realtime - database driven dialplan   bala krishnan
• [asterisk-users] Polycom not updating the directory list   hin lee
• [asterisk-users] Problems with Fax over TDM410P   hin lee
• [asterisk-users] How to set up Fax on Asterisk - Using analog Fax machines and HT502 (or FXS of a Digium TDM410P)   hin lee
• [asterisk-users] get hold event   bhrugu mehta
• [asterisk-users] VOIP Monitoring tools........   mike mosier
• [asterisk-users] VoIP monitoring tools   mike mosier
• [asterisk-users] RTCP How to stop   nakaji
• [asterisk-users] RTCP How to stop   nakaji
• [asterisk-users] problems originating an outgoing IAX2 call   nik600
• [asterisk-users] problems originating an outgoing IAX2 call   nik600
• [asterisk-users] Strange Centos Problem with Dahdi installation   russian qwerty
• [asterisk-users] Jitter Buffer and MeetMe.   russian qwerty
• [asterisk-users] Jitter Buffer and MeetMe.   russian qwerty
• [asterisk-users] kamailio   ram
• [asterisk-users] Asterisk Query   garge rama
• [asterisk-users] Caller ID on Asterisk and Astribank   frangky robert
• [asterisk-users] Outgoing routes with two PRI   chima s
• [asterisk-users] Hans Rauser   amit salunkhe
• [asterisk-users] Michael Wegner   mir shahnawaz
• [asterisk-users] Asterisk Load Balancing with Redfone/TDMoE driver   asterisk card support
• [asterisk-users] Dropped Calls   asterisk card support
• [asterisk-users] chan_ss7 issue   asterisk card support
• [asterisk-users] How set debug file for RxFax application   khalid touati
• [asterisk-users] How set debug file for RxFax application   khalid touati
• [asterisk-users] How set debug file for RxFax application   khalid touati
• [asterisk-users] Continuous bothering message -- Remote UNIX connection disconnected   khalid touati
• [asterisk-users] Continuous bothering message -- Remote UNIX connection disconnected   khalid touati
• [asterisk-users] How set debug file for RxFax application   khalid touati
• [asterisk-users] Continuous bothering message -- Remote UNIX connection disconnected   khalid touati
• [asterisk-users] How set debug file for RxFax application   khalid touati
• [asterisk-users] Continuous bothering message -- Remote UNIXconnection disconnected   khalid touati
• [asterisk-users] How set debug file for RxFax application   khalid touati
• [asterisk-users] Time variables in system application   khalid touati
• [asterisk-users] iptables miss up phone calls if not used properly   khalid touati
• [asterisk-users] Time variables in system application   khalid touati
• [asterisk-users] Time variables in system application   khalid touati
• [asterisk-users] iptables miss up phone calls if not used properly   khalid touati
• [asterisk-users] Interpbx connection   khalid touati
• [asterisk-users] Interpbx connection   khalid touati
• [asterisk-users] Interpbx connection   khalid touati
• [asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!   khalid touati
• [asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!   khalid touati
• [asterisk-users] Why app_fax.so there is no in asterisk16-1.6.2.6-1_centos5.x86_64.rpm?   Самусенко Андрей
• [asterisk-users] Why app_fax.so there is no in asterisk16-1.6.2.6-1_centos5.x86_64.rpm?   Самусенко Андрей

Last message date: Fri Apr 30 19:56:23 CDT 2010
Archived on: Fri Apr 30 19:56:31 CDT 2010

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