November 2007 Archives by author

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Starting: Thu Nov 1 00:48:57 CDT 2007
Ending: Fri Nov 30 23:42:43 CST 2007
Messages: 1778

• [asterisk-users] Asterisk version survey   Darrick Hartman (lists)
• [asterisk-users] Finding the status of an extension   Yehavi Bourvine +972-8-9489444
• [asterisk-users] retrieve last number dialled   Yehavi Bourvine +972-8-9489444
• [asterisk-users] Asterisk Sound File   Abdul
• [asterisk-users] Suppressing certain queue announcement voiceprompts   Torbjörn Abrahamsson
• [asterisk-users] DST   Joe Acquisto
• [asterisk-users] DST   Joe Acquisto
• [asterisk-users] DST   Joe Acquisto
• [asterisk-users] Sip to ATA?   Joe Acquisto
• [asterisk-users] Sip to ATA?   Joe Acquisto
• [asterisk-users] Dynamic Queue Members - Auto Logoff   Jason Adams
• [asterisk-users] Newb Question   Jeff Adams
• [asterisk-users] Newb Question   Jeff Adams
• [asterisk-users] Asterisk-Users: Termination   Mark Adams
• [asterisk-users] Asterisk ignoring manager events when busy   Nick Adams
• [asterisk-users] Asterisk ignoring manager events when busy   Nick Adams
• [asterisk-users] Interface with NEC NEAX 2400   Phil Allred
• [asterisk-users] Mobile phone codecs ...   Benny Amorsen
• [asterisk-users] Kirk IP600/3 Wireless Server SIP config   Benny Amorsen
• [asterisk-users] Kernel Native PCIE Network Cards?   Benny Amorsen
• [asterisk-users] Gigaset S450ip and simultaneous calls   Benny Amorsen
• [asterisk-users] Asterisk Sound File   Benny Amorsen
• [asterisk-users] Calling with hidden callerid   Benny Amorsen
• [asterisk-users] Calling with hidden callerid   Benny Amorsen
• [asterisk-users] Recommendations for 100 Wifi SIP phone setup   Benny Amorsen
• [asterisk-users] Asterisk behind a PIX firewall?   Benny Amorsen
• [asterisk-users] To DB or not to DB?   Benny Amorsen
• [asterisk-users] Outgoing PRI CID?   Anciso, Roy
• [asterisk-users] Selecting OSLEC for zaptel-1.4.6   Anciso, Roy
• [asterisk-users] Selecting OSLEC for zaptel-1.4.6   Anciso, Roy
• [asterisk-users] Cisco IP Communicator with Asterisk   Anciso, Roy
• [asterisk-users] Cisco IP Communicator with Asterisk   Anciso, Roy
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Anciso, Roy
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Anciso, Roy
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Anciso, Roy
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Anciso, Roy
• [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files   Anciso, Roy
• [asterisk-users] Page Command   Anciso, Roy
• [asterisk-users] Cisco phones and 32 directory object limit   Anciso, Roy
• [asterisk-users] Music on Hold Problem w/ Transfers   Anciso, Roy
• [asterisk-users] Music on Hold Problem w/ Transfers   Anciso, Roy
• [asterisk-users] Digium TE120P versus Sangoma A101D-X   Anciso, Roy
• [asterisk-users] Two PRI setup questions   Erik Anderson
• [asterisk-users] What is wrong with this mailing list   Erik Anderson
• [asterisk-users] What is wrong with this mailing list   Erik Anderson
• [asterisk-users] asterisk-stat problem   Erik Anderson
• [asterisk-users] OT - best policy for logs   Erik Anderson
• [asterisk-users] p2p t1 with sangoma hw   Erik Anderson
• [asterisk-users] OT: Best firmware for Linksys Router that is "SIP AWARE"   Erik Anderson
• [asterisk-users] OT: Best firmware for Linksys Router that is "SIP AWARE"   Erik Anderson
• [asterisk-users] Filesharing + video + voice supported Soft phone   Erik Anderson
• [asterisk-users] Urgent question.   Erik Anderson
• [asterisk-users] OT: Best firmware for Linksys Router thatis "SIP AWARE"   Erik Anderson
• [asterisk-users] Sangoma Question   Erik Anderson
• [asterisk-users] Switchvox Space Requirements   Andres
• [asterisk-users] sangoma zaptel patches   Andres
• [asterisk-users] __sip_xmit problem   Rilawich Ango
• [asterisk-users] zaptel.conf missing   Rilawich Ango
• [asterisk-users] crash   Rilawich Ango
• [asterisk-users] __sip_xmit problem   Rilawich Ango
• [asterisk-users] function voicemailmain   Rilawich Ango
• [asterisk-users] function voicemailmain   Rilawich Ango
• [asterisk-users] dtmf detection   Rilawich Ango
• [asterisk-users] DTMF Problem   Rilawich Ango
• [asterisk-users] quality after call transfer   Rilawich Ango
• [asterisk-users] Dial problem   Rilawich Ango
• [asterisk-users] dial in group   Rilawich Ango
• [asterisk-users] dial in group   Rilawich Ango
• [asterisk-users] MediaHandling--Help Required   srinivas Antarvedi
• [asterisk-users] Dial application response code--help required   srinivas Antarvedi
• [asterisk-users] ACD functionality , Skills for agents   Örn Arnarson
• [asterisk-users] Recommendations for 100 Wifi SIP phone setup   Julio Arruda
• [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4   Julio Arruda
• [asterisk-users] least cost routing and asterisk-1.4   Goke Aruna
• [asterisk-users] OT - best policy for logs   Jay R. Ashworth
• [asterisk-users] SIP Channels   Asterisk
• [asterisk-users] Asterisk SIP Channels Bridge   Asterisk
• [asterisk-users] Asterisk SIP Channels Bridge   Asterisk
• [asterisk-users] detect asterisk pbx via sip   Giedrius Augys
• [asterisk-users] detect asterisk pbx via sip   Giedrius Augys
• [asterisk-users] detect asterisk pbx via sip   Giedrius Augys
• [asterisk-users] Polycom SoundStation VTX 1000 with Asterisk?   Alvin Austin
• [asterisk-users] Polycom IP601 (mac)-directory.xml changes don't update phone   Alvin Austin
• [asterisk-users] Polycom IP601 call parking   Alvin Austin
• [asterisk-users] Call Parking/Pickup on a single button   Alvin Austin
• [asterisk-users] Client lost on skinny   Dan Austin
• [asterisk-users] Client lost on skinny   Dan Austin
• [asterisk-users] Issues after upgrading from 1.2 to 1.4: hangup immediately   Jean-Yves Avenard
• [asterisk-users] Issues after upgrading from 1.2 to 1.4: hangup immediately   Jean-Yves Avenard
• [asterisk-users] Issues after upgrading from 1.2 to 1.4: hangup immediately   Jean-Yves Avenard
• [asterisk-users] F3000 not connecting to Netgear routing   Dovid B
• [asterisk-users] Fax Problems with SpanDSP   Dovid B
• [asterisk-users] issues with downloads.digium.com   Dovid B
• [asterisk-users] F3000 not connecting to Netgear router   Dovid B
• [asterisk-users] Asterisk as a SIP to SIP Gateway   Dovid B
• [asterisk-users] Asterisk versions and H323   Dovid B
• [asterisk-users] Autodialing   Dovid B
• [asterisk-users] __sip_xmit problem   Dovid B
• [asterisk-users] sangoma zaptel patches   Dovid B
• [asterisk-users] 'Traditional' Faxing   Dovid B
• [asterisk-users] asterisk and installing chan_h323.so rpm   Dovid B
• [asterisk-users] Asterisk direct dialing   Dovid B
• [asterisk-users] Help: Static and dropped calls   Dovid B
• [asterisk-users] How to delete voice mail messages?   Dovid B
• [asterisk-users] Pass CallerID when call forwards to PSTN?   Dovid B
• [asterisk-users] multiple PBXs in one box   Dovid B
• [asterisk-users] Not Hearing hello-world Play   Dovid B
• [asterisk-users] Page Command   Dovid B
• [asterisk-users] SIP - ooh323 Bridging   Dovid B
• [asterisk-users] sidetone   Dovid B
• [asterisk-users] How to integrate Asterisk with Avaya   Dovid B
• [asterisk-users] RealTime register for iax DID   Dovid B
• [asterisk-users] H323 registeration and routing the calls   Dovid B
• [asterisk-users] OT Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.   Dovid B
• [asterisk-users] Problem installing Asterisk   Dovid B
• [asterisk-users] OT: Best firmware for Linksys Router that is "SIP AWARE"   Dovid B
• [asterisk-users] OT: Best firmware for Linksys Router thatis "SIP AWARE"   Dovid B
• [asterisk-users] Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.   Dovid B
• [asterisk-users] Hylafax   Dovid B
• [asterisk-users] Video Call   Marek B
• [asterisk-users] Video Call   Marek B
• [asterisk-users] issues with downloads.digium.com   Michiel van Baak
• [asterisk-users] How to delete voice mail messages?   Michiel van Baak
• [asterisk-users] What do you do to keep asterisk alive?   Chris Bagnall
• [asterisk-users] Asterisk on multi-homed systems   Chris Bagnall
• [asterisk-users] Asterisk on multi-homed systems   Chris Bagnall
• [asterisk-users] Which files to be copied   Alex Balashov
• [asterisk-users] Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.   Alex Balashov
• [asterisk-users] Annoying PRI Channels Restarting Message   Alex Balashov
• [asterisk-users] Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.   Alex Balashov
• [asterisk-users] Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.   Alex Balashov
• [asterisk-users] Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.   Alex Balashov
• [asterisk-users] Urgent question.   Alex Balashov
• [asterisk-users] Urgent question.   Alex Balashov
• [asterisk-users] Urgent question.   Alex Balashov
• [asterisk-users] SIP port 5060 closed - how do I open it?   Alex Balashov
• [asterisk-users] least cost routing and asterisk-1.4   Alex Balashov
• [asterisk-users] least cost routing and asterisk-1.4   Alex Balashov
• [asterisk-users] Protection switching on PRIs.   Alex Balashov
• [asterisk-users] Only call me once   Alex Balashov
• [asterisk-users] How to setup redundant SIP peers   Thomas Balsfulland
• [asterisk-users] Wifi handover/roaming   Luis Antonio Prata Barbosa
• [asterisk-users] Kirk IP600/3 Wireless Server SIP config   Luis Antonio Prata Barbosa
• [asterisk-users] wifi   Luis Antonio Prata Barbosa
• [asterisk-users] r2 multiframe error - continue   Luis Antonio Prata Barbosa
• [asterisk-users] Kirk IP600/3 Wireless Server SIP config   Remco Barendse
• [asterisk-users] Asterisk on Pcengines Alix board   Giuseppe Barichello
• [asterisk-users] Asterisk on Pcengines Alix board   Giuseppe Barichello
• [asterisk-users] Asterisk on Pcengines Alix board   Giuseppe Barichello
• [asterisk-users] No sound from playback and voicemail   Forrest Beck
• [asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card - Update   Joseph Begumisa
• [asterisk-users] Asterisk and OBDC   Maxi Belino
• [asterisk-users] Internal CallerID problem   Mark Bell
• [asterisk-users] Caller ID Question   Mark Bell
• [asterisk-users] Problem installing Asterisk   Ugo Bellavance
• [asterisk-users] NAT keep-alive   Ugo Bellavance
• [asterisk-users] NAT keep-alive   Ugo Bellavance
• [asterisk-users] NAT keep-alive   Ugo Bellavance
• [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4   Ugo Bellavance
• [asterisk-users] Asterisk behind a PIX firewall?   Ugo Bellavance
• [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4   Fernando Berretta
• [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4   Fernando Berretta
• [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4   Fernando Berretta
• [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4   Fernando Berretta
• [asterisk-users] svn+chan_mobile+Asterisk+Siemens FXO no voice   Mandeep Singh Bhabha
• [asterisk-users] Mystery phone!   Mandeep Singh Bhabha
• [asterisk-users] Switch to Multi-Proc -> Choppy sound?   Bill Binko
• [asterisk-users] Switch to Multi-Proc -> Choppy sound?   Bill Binko
• [asterisk-users] Asterisk & Cisco calling Name   John Bittner
• [asterisk-users] SIP Trunk Problems   Nicholas Blasgen
• [asterisk-users] Problem with AGI Script   Chris Blunt
• [asterisk-users] 1.4 SIP Jitter Buffer   Gregory Boehnlein
• [asterisk-users] 1.4 SIP Jitter Buffer   Gregory Boehnlein
• [asterisk-users] Digium Vs sangoma Hradware   Stephen Bosch
• [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID   Stephen Bosch
• [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID   Stephen Bosch
• [asterisk-users] Are the ATAs which can allow multiple extensions from one network connection?   Dave Bour
• [asterisk-users] asterisk and installing chan_h323.so rpm   David Boyd
• [asterisk-users] OT: Best firmware for Linksys Router that is "SIP AWARE"   David Boyd
• [asterisk-users] AEL2 and Callbacks   Sean Bright
• [asterisk-users] AEL2 and Callbacks   Sean Bright
• [asterisk-users] How to play Asterisk .raw file   Sean Bright
• [asterisk-users] 7960 Queue Issue   Nick Brown
• [asterisk-users] 7960 Queue Issue   Nick Brown
• [asterisk-users] Dynamic Queue Members - Auto Logoff   Nick Brown
• [asterisk-users] MOH Codec Issue   Nick Brown
• [asterisk-users] MOH Codec Issue   Nick Brown
• [asterisk-users] MOH Codec Issue - Fixed   Nick Brown
• [asterisk-users] Music on Hold -- Error   Nick Brown
• [asterisk-users] Calling with hidden callerid   Nick Brown
• [asterisk-users] SIP port 5060 closed - how do I open it?   Nick Brown
• [asterisk-users] Simple Asterisk to Asterisk SIP Call Setup?   Russell Brown
• [asterisk-users] Shared line appearance phones?   Russell Bryant
• [asterisk-users] Shared line appearance phones?   Russell Bryant
• [asterisk-users] SLA: Handling of errors in outgoing call   Russell Bryant
• [asterisk-users] Shared line appearance phones?   Russell Bryant
• [asterisk-users] 3pcc/click to dial accounting   CSB
• [asterisk-users] Extracting custom headers from SIP REFER   CSB
• [asterisk-users] Determination of billsec   CSB
• [asterisk-users] Determination of billsec   CSB
• [asterisk-users] Determination of billsec   CSB
• [asterisk-users] Determination of billsec   CSB
• [asterisk-users] asterisk-stat problem   Richard Cahilig
• [asterisk-users] Asterisk Recording   Richard Cahilig
• [asterisk-users] Asterisk Recording   Richard Cahilig
• [asterisk-users] AsteriskNOW and TDM800P   Rafael Canchola
• [asterisk-users] Problem receiving audio with Asterisk 1.4.5 with SIP trunk   Rafael Canchola
• [asterisk-users] Registration problem: UA -> SER -> Asterisk   Stefano Capitanio
• [asterisk-users] Registration problem: UA -> SER -> Asterisk   Stefano Capitanio
• [asterisk-users] Conference rooms   Fabio Cappelletti
• [asterisk-users] Natural Microsystems AG Quad   Marco Carvalho
• [asterisk-users] Realtime extensions configuration - calling user filtering   Ricardo Carvalho
• [asterisk-users] Realtime extensions configuration - calling user filtering   Ricardo Carvalho
• [asterisk-users] Softphone to be installed on the Mobile   Ricardo Carvalho
• [asterisk-users] Get IP address of an incoming or outgoing SIP call   Ricardo Carvalho
• [asterisk-users] Asterisk behind a PIX firewall?   Ricardo Carvalho
• [asterisk-users] Finding the status of an extension   Ricardo Carvalho
• [asterisk-users] hostname in MySQL CDR records   Ricardo Carvalho
• [asterisk-users] Shared line appearance phones?   Ricardo Carvalho
• [asterisk-users] Do While loop   Ricardo Carvalho
• [asterisk-users] Audiocodes over Sat link. and delay   Justin Case
• [asterisk-users] Internal CallerID problem   Justin Case
• [asterisk-users] Caller ID Question   Justin Case
• [asterisk-users] Route an incoming call by ANI*DNIS   Dan Casey
• [asterisk-users] Route an incoming call by ANI*DNIS   Dan Casey
• [asterisk-users] Route an incoming call by ANI*DNIS   Dan Casey
• [asterisk-users] Route an incoming call by ANI*DNIS   Dan Casey
• [asterisk-users] Asterisk as a SIP to XMPP Jingle voice gateway   Eric Chamberlain
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Eric Chamberlain
• [asterisk-users] Asterisk as a SIP to XMPP Jingle voice gateway   Eric Chamberlain
• [asterisk-users] California based PSTN connections   Eric Chamberlain
• [asterisk-users] Softphone to be installed on the Mobile   Eric Chamberlain
• [asterisk-users] sip proxy failover   Eric Chamberlain
• [asterisk-users] Changing audio message to text message   Anthony Chapellier
• [asterisk-users] Changing audio message to text message   Anthony Chapellier
• [asterisk-users] Changing audio message to text message   Anthony Chapellier
• [asterisk-users] Changing audio message to text message   Anthony Chapellier
• [asterisk-users] Digium and Asterisk   Anthony Chapellier
• [asterisk-users] Asterisk API Manager   Anthony Chapellier
• [asterisk-users] Asterisk API Manager   Anthony Chapellier
• [asterisk-users] Asterisk API Manager   Anthony Chapellier
• [asterisk-users] Problem with CDR userfield not being set   Carlos Chavez
• [asterisk-users] PRI dialout problem with some numbers...   Carlos Chavez
• [asterisk-users] No sound from playback and voicemail   Carlos Chavez
• [asterisk-users] RTP traffic not being forwarded   Carlos Chavez
• [asterisk-users] Problem dialing certain numbers with an E1 PRI   Carlos Chavez
• [asterisk-users] Asterisk as a SIP to XMPP Jingle voice gateway   Mik Cheez
• [asterisk-users] IAX complaints? What are they?   Dr. Michael J. Chudobiak
• [asterisk-users] Are the ATAs which can allow multiple extensions from one network connection?   Frank Church
• [asterisk-users] Problems getting Asterisk to detect call in SUSE9.3, with FritzCard   Frank Church
• [asterisk-users] Problems getting Asterisk to detect call in SUSE9.3, with FritzCard   Frank Church
• [asterisk-users] Problems getting Asterisk to detect call in SUSE9.3, with FritzCard   Frank Church
• [asterisk-users] 'Traditional' Faxing   Greg Cockburn
• [asterisk-users] PRI commands missing...   Tzafrir Cohen
• [asterisk-users] zaptel.conf missing   Tzafrir Cohen
• [asterisk-users] ztdummy and BackGround   Tzafrir Cohen
• [asterisk-users] Issues after upgrading from 1.2 to 1.4: hangup immediately   Tzafrir Cohen
• [asterisk-users] TE220 PCI express performnace   Tzafrir Cohen
• [asterisk-users] Asterisk & OpenVZ   Tzafrir Cohen
• [asterisk-users] Sangoma S200 and Digium TDM400P together   Tzafrir Cohen
• [asterisk-users] Selecting OSLEC for zaptel-1.4.6   Tzafrir Cohen
• [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking   Tzafrir Cohen
• [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking   Tzafrir Cohen
• [asterisk-users] What do you do to keep asterisk alive?   Tzafrir Cohen
• [asterisk-users] What do you do to keep asterisk alive?   Tzafrir Cohen
• [asterisk-users] ztdummy, zttest   Tzafrir Cohen
• [asterisk-users] Kernel Native PCIE Network Cards?   Tzafrir Cohen
• [asterisk-users] mpg123 on Thecus N2100   Tzafrir Cohen
• [asterisk-users] asterisk 1.4 prereq   Tzafrir Cohen
• [asterisk-users] sangoma asterisk patches   Tzafrir Cohen
• [asterisk-users] sangoma zaptel patches   Tzafrir Cohen
• [asterisk-users] sangoma zaptel patches   Tzafrir Cohen
• [asterisk-users] [not about sangoma/zaptel] Playback() clicking sound at the end of the prompt   Tzafrir Cohen
• [asterisk-users] ztdummy, zttest   Tzafrir Cohen
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Tzafrir Cohen
• [asterisk-users] No sound from playback and voicemail   Tzafrir Cohen
• [asterisk-users] Stress-Testing Asterisk   Tzafrir Cohen
• [asterisk-users] Install Scripts for CentOS 4   Tzafrir Cohen
• [asterisk-users] function voicemailmain   Tzafrir Cohen
• [asterisk-users] make config update-rc.d   Tzafrir Cohen
• [asterisk-users] problem with tdm2400p configuration   Tzafrir Cohen
• [asterisk-users] problem with tdm2400p configuration   Tzafrir Cohen
• [asterisk-users] problem with tdm2400p configuration   Tzafrir Cohen
• [asterisk-users] Conference Call Dial-Out to a participant   Tzafrir Cohen
• [asterisk-users] asterisk as non-root/best practices   Tzafrir Cohen
• [asterisk-users] Reporting bugs   Tzafrir Cohen
• [asterisk-users] FXO Hangs up automatically   Tzafrir Cohen
• [asterisk-users] Zaptel 1.4 spec file   Tzafrir Cohen
• [asterisk-users] Zaptel 1.4 spec file   Tzafrir Cohen
• [asterisk-users] asterisk as non-root/best practices   Tzafrir Cohen
• [asterisk-users] asterisk as non-root/best practices   Tzafrir Cohen
• [asterisk-users] TDM808B 8 port FXO setting problem   Tzafrir Cohen
• [asterisk-users] Digium E1 and Digium TDM400P (2xFXO) Help!   Tzafrir Cohen
• [asterisk-users] Possible Conflicts with Junghanns 4 Port BRI and 8 Port Sangoma Analog in Same Box?   Tzafrir Cohen
• [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4   Tzafrir Cohen
• [asterisk-users] Copy or Make + Make Install   Tzafrir Cohen
• [asterisk-users] Fw: Remove a TDM Card   Tzafrir Cohen
• [asterisk-users] Fw: Remove a TDM Card   Tzafrir Cohen
• [asterisk-users] Fw: Remove a TDM Card   Tzafrir Cohen
• [asterisk-users] Fw: Remove a TDM Card   Tzafrir Cohen
• [asterisk-users] Asterisk Program Closes   Ryan M. Colbert
• [asterisk-users] Music on Hold -- Error   Ryan M. Colbert
• [asterisk-users] Music on Hold -- Error   Ryan M. Colbert
• [asterisk-users] Druid   Dean Collins
• [asterisk-users] Druid   Dean Collins
• [asterisk-users] Druid   Dean Collins
• [asterisk-users] Druid   Dean Collins
• [asterisk-users] Softphone to be installed on the Mobile   Dean Collins
• [asterisk-users] Free T1 Card?   Michael Collins
• [asterisk-users] Free T1 Card?   Michael Collins
• [asterisk-users] Interface with NEC NEAX 2400   Michael Collins
• [asterisk-users] Annoying PRI Channels Restarting Message   Michael Collins
• [asterisk-users] Bad audio quality in 1.4-SVN when encoding to alaw/ulaw   Joshua Colp
• [asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX   John Constalgie
• [asterisk-users] Finding the status of an extension   Andre Courchesne - Consultant
• [asterisk-users] TDM808B 8 port FXO setting problem   Gustavo Cordeiro
• [asterisk-users] Using php exec() in agi script   Andre Courchesne
• [asterisk-users] Problem with AGI Script   Brett Crapser
• [asterisk-users] Conference Call Dial-Out to a participant   Brett Crapser
• [asterisk-users] Phones Not Registering   Brett Crapser
• [asterisk-users] Caller ID Question   CunningPike
• [asterisk-users] Voice mail & Uniden UIP-200 phones   CunningPike
• [asterisk-users] asterisk as non-root/best practices   CunningPike
• [asterisk-users] asterisk as non-root/best practices   CunningPike
• [asterisk-users] To DB or not to DB?   Matt Darnell
• [asterisk-users] Need Reference sites   Michael Davidson
• [asterisk-users] RTP Read too short   Steve Davies
• [asterisk-users] Linksys SPA-941 Unavailable   Steve Davies
• [asterisk-users] __sip_xmit problem   Steve Davies
• [asterisk-users] __sip_xmit problem   Steve Davies
• [asterisk-users] Attended transfer to Queue   Steve Davies
• [asterisk-users] Snom phones, blinking lights and call pickup   Steve Davies
• [asterisk-users] Snom phones, blinking lights and call pickup   Steve Davies
• [asterisk-users] Problems with losing D-Channel on   Eric Delaporte
• [asterisk-users] OT: Snom 300 losing config?   Mike Dent
• [asterisk-users] Blackberry MVS and Asterisk.   Mike Dent
• [asterisk-users] Agi manager session.   Voip Dev
• [asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered   Costa Dinoteli
• [asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered   Costa Dinoteli
• [asterisk-users] How to integrate Asterisk with Avaya   Costa Dinoteli
• [asterisk-users] Intercom with Snom phones   Doug
• [asterisk-users] Asterisk Help   Doug
• [asterisk-users] Determination of billsec   Doug
• [asterisk-users] Polycom Speakerphone   Doug
• [asterisk-users] Polycom Speakerphone   Doug
• [asterisk-users] DTMF Problem   Doug
• [asterisk-users] Asterisk Program Closes   Doug
• [asterisk-users] Page Command   Doug
• [asterisk-users] detecting voltage on fxo   Paradise Dove
• [asterisk-users] Asterisk behind a PIX firewall?   Shlomo Dubrowin
• [asterisk-users] Meetme - how to protect the conference?   Darryl Dunkin
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Darryl Dunkin
• [asterisk-users] G729/MOH Quality   Darryl Dunkin
• [asterisk-users] libpri & tie line vs trunk   Michelle Dupuis
• [asterisk-users] PRI over T1 calls dropping, cause 100   Michelle Dupuis
• [asterisk-users] OT: Aastra 57i configuration via TFTP problem   Michelle Dupuis
• [asterisk-users] OT: Aastra 57i configuration via TFTP problem   Michelle Dupuis
• [asterisk-users] OT: Aastra 57i configuration via TFTP problem   Michelle Dupuis
• [asterisk-users] If caller id is null set to a specific number   Michelle Dupuis
• [asterisk-users] Nortel digital FXO channel bank? Exists?   Michelle Dupuis
• [asterisk-users] How to pay for libpri development   Michelle Dupuis
• [asterisk-users] Nortel digital FXO channel bank? Exists?   Michelle Dupuis
• [asterisk-users] facilityenable in zapata.conf   Michelle Dupuis
• [asterisk-users] Aastra 480i CT - No Incoming Calls   Michelle Dupuis
• [asterisk-users] Reporting bugs   Robert Dyck
• [asterisk-users] Extracting custom headers from SIP REFER   Johansson Olle E
• [asterisk-users] SIP: "To:" header?   Johansson Olle E
• [asterisk-users] __sip_xmit problem   Johansson Olle E
• [asterisk-users] Asterisk 1.4 + Presence   Johansson Olle E
• [asterisk-users] SIP: "To:" header?   Johansson Olle E
• [asterisk-users] Call Forward on SIP unreachable (network failure)   Johansson Olle E
• [asterisk-users] route INVITE sip:s at sip.test.fr   Johansson Olle E
• [asterisk-users] Help on strange problem...   Johansson Olle E
• [asterisk-users] channels to destroy   Johansson Olle E
• [asterisk-users] reload command   Johansson Olle E
• [asterisk-users] Help: How to configure SIP domain on SPA942   Johansson Olle E
• [asterisk-users] Changing audio message to text message   EdPimentl
• [asterisk-users] Queue Statistics Reporting   Russell Eden
• [asterisk-users] hostname in MySQL CDR records   Steve Edwards
• [asterisk-users] use dial plan passed arg value in C agi code   Steve Edwards
• [asterisk-users] MeetMe CPU resources   Steve Edwards
• [asterisk-users] MeetMe CPU resources   Steve Edwards
• [asterisk-users] use dial plan passed arg value in C agi code   Steve Edwards
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Steve Edwards
• [asterisk-users] Asterisk on Zonbu, impact of transcoding   Steve Edwards
• [asterisk-users] Problem with AGI Script   Steve Edwards
• [asterisk-users] Problem with AGI Script   Steve Edwards
• [asterisk-users] OT - best policy for logs   Steve Edwards
• [asterisk-users] Dumb AGI question   Steve Edwards
• [asterisk-users] Broadcast dialing/playback   Steve Edwards
• [asterisk-users] Billing/Call Control engine : AGI scripts/ AstMan API   Steve Edwards
• [asterisk-users] Asterisk API Manager   Steve Edwards
• [asterisk-users] Newb Question   Steve Edwards
• [asterisk-users] asterisk as non-root/best practices   Steve Edwards
• [asterisk-users] AEX800 (TDM800 Express) - not detected   Mark J Elkins
• [asterisk-users] Druid   Alex Epshteyn
• [asterisk-users] Druid   Alex Epshteyn
• [asterisk-users] MeetMe CPU resources   Carles Pina i Estany
• [asterisk-users] MeetMe CPU resources   Carles Pina i Estany
• [asterisk-users] MeetMe CPU resources   Carles Pina i Estany
• [asterisk-users] ztdummy, zttest   Carles Pina i Estany
• [asterisk-users] ztdummy, zttest   Carles Pina i Estany
• [asterisk-users] ztdummy, zttest   Carles Pina i Estany
• [asterisk-users] ztdummy, zttest   Carles Pina i Estany
• [asterisk-users] ztdummy and BackGround   Carles Pina i Estany
• [asterisk-users] ztdummy, zttest   Carles Pina i Estany
• [asterisk-users] channels to destroy   Carles Pina i Estany
• [asterisk-users] channels to destroy   Carles Pina i Estany
• [asterisk-users] not sending bye   Carles Pina i Estany
• [asterisk-users] Which Variable???   C F
• [asterisk-users] What do you do to keep asterisk alive?   C F
• [asterisk-users] asterisk integration with panasonic analog pbx   C F
• [asterisk-users] How to integrate Asterisk with Avaya   C F
• [asterisk-users] Broadcast dialing/playback   C F
• [asterisk-users] Asterisk behind a PIX firewall?   C F
• [asterisk-users] To DB or not to DB?   C F
• [asterisk-users] Polycom Speakerphone   F6HQZ
• [asterisk-users] quality after call transfer   F6HQZ
• [asterisk-users] TDM808B 8 port FXO setting problem   F6HQZ
• [asterisk-users] Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ?   Don Fanning
• [asterisk-users] Linksys 942 Call Transfer   Jon Farmer
• [asterisk-users] RTP Read too short   John Faubion
• [asterisk-users] RTP Read too short   John Faubion
• [asterisk-users] Grandstream GXP2020 + Asterisk 1.4.11   John Faubion
• [asterisk-users] Asterisk on Pcengines Alix board   John Faubion
• [asterisk-users] Asterisk on Pcengines Alix board   John Faubion
• [asterisk-users] Asterisk on Pcengines Alix board   John Faubion
• [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking   Hans Feringa
• [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking   Hans Feringa
• [asterisk-users] Asterisk and OBDC   Gilberto Nunes Ferreira
• [asterisk-users] Asterisk version survey   Leonardo Gomes Figueira
• [asterisk-users] use dial plan passed arg value in C agi code   James FitzGibbon
• [asterisk-users] use dial plan passed arg value in C agi code   James FitzGibbon
• [asterisk-users] Dynamic Queue Members - Auto Logoff   James FitzGibbon
• [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID   James FitzGibbon
• [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID   James FitzGibbon
• [asterisk-users] 'a' extension   James FitzGibbon
• [asterisk-users] ACD Queue LOG RINGNOANSWER Content 0   James FitzGibbon
• [asterisk-users] ACD functionality , Skills for agents   James FitzGibbon
• [asterisk-users] DST   Kevin P. Fleming
• [asterisk-users] AEX800 (TDM800 Express) - not detected   Kevin P. Fleming
• [asterisk-users] Switchvox Space Requirements   Kevin P. Fleming
• [asterisk-users] Multiple B410P's in one machine   Kevin P. Fleming
• [asterisk-users] Digium TE120P versus Sangoma A101D-X   Kevin P. Fleming
• [asterisk-users] Digium TE120P versus Sangoma A101D-X   Kevin P. Fleming
• [asterisk-users] Druid   Anthony Francis
• [asterisk-users] What is wrong with this mailing list   Anthony Francis
• [asterisk-users] Asterisk 1.4 with LDAP   Anthony Francis
• [asterisk-users] [asterisk-dev] trunk working under windows!   Anthony Francis
• [asterisk-users] Problems with losing D-Channel on   Anthony Francis
• [asterisk-users] To DB or not to DB?   Anthony Francis
• [asterisk-users] PRI over T1 calls dropping, cause 100   Matthew Fredrickson
• [asterisk-users] PRI commands missing...   Matthew Fredrickson
• [asterisk-users] issues with downloads.digium.com   Matthew Fredrickson
• [asterisk-users] Two PRI setup questions   Matthew Fredrickson
• [asterisk-users] Please explain the correct LED color for B410P   Matthew Fredrickson
• [asterisk-users] sangoma zaptel patches   Matthew Fredrickson
• [asterisk-users] Two B-Channel Transfer (2BCT/TBCT) Trobule on DMS100 PRI   Matthew Fredrickson
• [asterisk-users] Problem dialing certain numbers with an E1 PRI   Matthew Fredrickson
• [asterisk-users] Digium TE120P versus Sangoma A101D-X   Matthew Fredrickson
• [asterisk-users] IAX complaints? What are they?   Matthew Fredrickson
• [asterisk-users] Outgoing PRI CID?   Turbo Fredriksson
• [asterisk-users] Outgoing PRI CID?   Turbo Fredriksson
• [asterisk-users] Outgoing PRI CID?   Turbo Fredriksson
• [asterisk-users] DST   Turbo Fredriksson
• [asterisk-users] PRI debuging shows 'Ext: 0' (Was: Outgoing PRI CID?)   Turbo Fredriksson
• [asterisk-users] Outgoing PRI CID?   Turbo Fredriksson
• [asterisk-users] Solved: Outgoing PRI CID?   Turbo Fredriksson
• [asterisk-users] Restart when convenient   Turbo Fredriksson
• [asterisk-users] Correct syntax for IF()?   Turbo Fredriksson
• [asterisk-users] PHP - Queues - etc.   Marcelo Freitas
• [asterisk-users] DST   Dave Fullerton
• [asterisk-users] Which Variable???   Dave Fullerton
• [asterisk-users] Selecting OSLEC for zaptel-1.4.6   Dave Fullerton
• [asterisk-users] 'Traditional' Faxing   Dave Fullerton
• [asterisk-users] Transfering IAX context   Dave Fullerton
• [asterisk-users] Fw: Remove a TDM Card   Dave Fullerton
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Andrew Furey
• [asterisk-users] Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.   GNUbie
• [asterisk-users] Sangoma S200 and Digium TDM400P together   Paulo Garcia
• [asterisk-users] AEL2 and Callbacks   Douglas Garstang
• [asterisk-users] Building an Asterisk 1.4 RPM.   Douglas Garstang
• [asterisk-users] Building an Asterisk 1.4 RPM   Douglas Garstang
• [asterisk-users] Zaptel 1.4 spec file   Douglas Garstang
• [asterisk-users] Multiple Return Values from func_odbc   Douglas Garstang
• [asterisk-users] Multiple Return Values from func_odbc   Douglas Garstang
• [asterisk-users] Multiple Return Values from func_odbc   Douglas Garstang
• [asterisk-users] How to play Asterisk .raw file   Gary
• [asterisk-users] sip show peers in 1.4.13   Jerry Geis
• [asterisk-users] Grandstream 488 and DTMF with a call file   Jerry Geis
• [asterisk-users] dtmfmode RFC2833 and inband   Jerry Geis
• [asterisk-users] time on polycom 501   Jerry Geis
• [asterisk-users] reload command   Jerry Geis
• [asterisk-users] reload command   Jerry Geis
• [asterisk-users] Interesting Conference Request - Can this be done ?   Bob Gibson
• [asterisk-users] Video Call   Bob Gibson
• [asterisk-users] Softphone to be installed on the Mobile   Bob Gibson
• [asterisk-users] VMukti - Filesharing + video + voice supported Soft phone   Bob Gibson
• [asterisk-users] Filesharing + video + voice supported Soft phone   Bob Gibson
• [asterisk-users] RTP Read too short   Drew Gibson
• [asterisk-users] [asterisk-dev] trunk working under windows!   Drew Gibson
• [asterisk-users] [Fwd: voicemail locked up Asterisk 1.4.13]   Lyle Giese
• [asterisk-users] Help: Asterisk info   Lyle Giese
• [asterisk-users] Voice mail & Uniden UIP-200 phones   Lyle Giese
• [asterisk-users] Voice mail & Uniden UIP-200 phones   Lyle Giese
• [asterisk-users] Using existing extensions.conf macros, and co-habitation   Lyle Giese
• [asterisk-users] Off-Topic: Avaya   Salvatore Giudice
• [asterisk-users] Outgoing PSTN calls , unusable voice quality   Salvatore Giudice
• [asterisk-users] ring group containing external 10-digit numbers   David Gomillion
• [asterisk-users] time on polycom 501   David Gomillion
• [asterisk-users] Polycom Speakerphone   David Gomillion
• [asterisk-users] Digium E1 and Digium TDM400P (2xFXO) Help!   David Gomillion
• [asterisk-users] Chatterbug   Robert Goodyear
• [asterisk-users] Chatterbug   Robert Goodyear
• [asterisk-users] wifi   Michael Graves
• [asterisk-users] wifi   Michael Graves
• [asterisk-users] Polycom Speakerphone   Michael Graves
• [asterisk-users] Recommendations for 100 Wifi SIP phone setup   Michael Graves
• [asterisk-users] Semi-OT: Best Speakerphone   Michael Graves
• [asterisk-users] OpenSER for Asterisk Load balance   Edgar Guadamuz
• [asterisk-users] OpenSER for Asterisk Load balance   Edgar Guadamuz
• [asterisk-users] Little OT: Compilation of EICON driver, fails with capi errors   Stefan Guenther
• [asterisk-users] No sound from playback and voicemail   Stefan Guenther
• [asterisk-users] No sound from playback and voicemail (Carlos Chavez)   Stefan Guenther
• [asterisk-users] No sound from playback and voicemail (Atis Lezdins)   Stefan Guenther
• [asterisk-users] Hylafax   Sahil Gupta
• [Asterisk-Users] Re: Unable to use voicemail(Please suggestion)   CW_ASN - Gus
• [asterisk-users] 7960 Queue Issue   Paul Hales
• [asterisk-users] Snom 320 with TDM02B and echo problems   Paul Hales
• [asterisk-users] MOH Codec Issue   Paul Hales
• [asterisk-users] Chatterbug   Paul Hales
• [asterisk-users] MOH Codec Issue   Paul Hales
• [asterisk-users] MOH Codec Issue - Fixed   Paul Hales
• [asterisk-users] asterisk-stat problem   Paul Hales
• [asterisk-users] Work   Paul Hales
• [asterisk-users] Calling with hidden callerid   Paul Hales
• [asterisk-users] DTMF not recognized on ISDN with Siemens -not IP- phone   Paul Hales
• [asterisk-users] Digium TE120P versus Sangoma A101D-X   Paul Hales
• [asterisk-users] Realtime SIP & BLF   Paul Hales
• [asterisk-users] G729 on wrong bus   Paul Hales
• [asterisk-users] Asterisk behind a PIX firewall?   Wendell Hamilton
• [asterisk-users] e911   Mike Hammett
• [asterisk-users] e911   Mike Hammett
• [asterisk-users] e911   Mike Hammett
• [asterisk-users] Help   Chris 'Xenon' Hanson
• [asterisk-users] Asterisk version survey   Darrick Hartman
• [asterisk-users] Video Call   Bert Haverkamp
• [asterisk-users] Odd bug in Siemens C460IP ?   Paul Hayes
• [asterisk-users] Realtime SIP & BLF   Daniel Hazelbaker
• [asterisk-users] Which Variable???   Gordon Henderson
• [asterisk-users] Video Call   Gordon Henderson
• [asterisk-users] Switchvox Space Requirements   Gordon Henderson
• [asterisk-users] Softphone to be installed on the Mobile   Gordon Henderson
• [asterisk-users] Phones Not Registering   Gordon Henderson
• [asterisk-users] Calling with hidden callerid   Gordon Henderson
• [asterisk-users] dial in group   Gordon Henderson
• [asterisk-users] dial in group   Gordon Henderson
• [asterisk-users] Asterisk behind a PIX firewall?   Gordon Henderson
• [asterisk-users] DTMF not recognized on ISDN with Siemens -not IP- phone   Gordon Henderson
• [asterisk-users] Polycom softkey transfer issue   Wayne P. Hill
• [asterisk-users] common/shared voicemail box   Rob Hillis
• [asterisk-users] Polycom Park Button   Lorentz Hinrichsen
• [asterisk-users] asterisk as a gateway   Rizwan Hisham
• [asterisk-users] __sip_xmit problem   Rizwan Hisham
• [asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered   Rizwan Hisham
• [asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered   Rizwan Hisham
• [asterisk-users] asterisk-users Digest, Vol 40, Issue 5   Rizwan Hisham
• [asterisk-users] Digium E1 and Digium TDM400P (2xFXO) Help!   Cameron Hissey
• [asterisk-users] Digium E1 and Digium TDM400P (2xFXO) Help!   Cameron Hissey
• [asterisk-users] issues with downloads.digium.com   Anselm Martin Hoffmeister
• [asterisk-users] Recording just first part of call?   Anselm Martin Hoffmeister
• [asterisk-users] VoiceMail hangup   Anselm Martin Hoffmeister
• [asterisk-users] [Fwd: Re: VoiceMail hangup]   Anselm Martin Hoffmeister
• [asterisk-users] Gigaset S450ip and simultaneous calls   Anselm Martin Hoffmeister
• [asterisk-users] Aastra 480i CT - No Incoming Calls   Anselm Martin Hoffmeister
• [asterisk-users] Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ?   Anselm Martin Hoffmeister
• [asterisk-users] Recommendations for 100 Wifi SIP phone setup   Anselm Martin Hoffmeister
• [asterisk-users] Pass CallerID when call forwards to PSTN?   Russell Horn
• [asterisk-users] Pass CallerID when call forwards to PSTN?   Russell Horn
• [asterisk-users] Pass CallerID when call forwards to PSTN?   Russell Horn
• [asterisk-users] Mystery phone!   Jim Houser
• [asterisk-users] Dialing time-out   Jim Houser
• [asterisk-users] Off-Topic: Avaya   Jim Houser
• [asterisk-users] Routing Anonymous Callerid   C. Duncan Hudson
• [asterisk-users] Asterisk on multi-homed systems   James Texter III
• [asterisk-users] Help: How to configure SIP domain on SPA942   Ira
• [asterisk-users] Calling with hidden callerid   Ira
• [asterisk-users] Asterisk version survey   Ira
• [asterisk-users] Sip to ATA?   Ira
• [asterisk-users] Call Parking/Pickup on a single button   Ira
• [asterisk-users] Video Call   Yann JOUANIN
• [asterisk-users] detect asterisk pbx via sip   Yann JOUANIN
• [asterisk-users] Conference Call Dial-Out to a participant   Yann JOUANIN
• [asterisk-users] Polycom Speakerphone   Eric Jacksch
• [asterisk-users] Problem with AGI Script   Benjamin Jacob
• [asterisk-users] DTMF Problem   Benjamin Jacob
• [asterisk-users] Problem with AGI Script   Benjamin Jacob
• [asterisk-users] common/shared voicemail box   Benjamin Jacob
• [asterisk-users] Phones Not Registering   Benjamin Jacob
• [asterisk-users] Billing/Call Control engine : AGI scripts/ AstMan API   Benjamin Jacob
• [asterisk-users] retrieve last number dialled   Benjamin Jacob
• [asterisk-users] retrieve last number dialled   Benjamin Jacob
• [asterisk-users] Asterisk as a SIP to XMPP Jingle voice gateway   Geoff Jacobs
• [asterisk-users] Help   Jarga Jallow
• [asterisk-users] Help   Jarga Jallow
• [asterisk-users] Help: Static and dropped calls   Jarga Jallow
• [asterisk-users] Help: Asterisk info   Jarga Jallow
• [asterisk-users] Asterisk Help   Jarga Jallow
• [asterisk-users] Asterisk Help   Jarga Jallow
• [asterisk-users] Help with Polycom 320   Jarga Jallow
• [asterisk-users] Help Dial extention   Jarga Jallow
• [asterisk-users] Cisco IP Communicator with Asterisk   Kai-Uwe Jensen
• [asterisk-users] Cisco IP Communicator with Asterisk   Kai-Uwe Jensen
• [asterisk-users] Arbitrary limit on length of email address?   Per Jessen
• [asterisk-users] grandstream troubles   Per Jessen
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] Linksys SPA-941 Unavailable   Per Jessen
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] grandstream troubles   Per Jessen
• [asterisk-users] What do you do to keep asterisk alive?   Per Jessen
• [asterisk-users] 'Traditional' Faxing   Per Jessen
• [asterisk-users] Change the Voice promps in asterisk 1.4   Per Jessen
• [asterisk-users] Building and running mISDN for B410P on Ubuntu 7.04   Per Jessen
• [asterisk-users] Gigaset S450ip and simultaneous calls   Per Jessen
• [asterisk-users] iptables requirements for SIP   Per Jessen
• [asterisk-users] Problems getting Asterisk to detect call in SUSE9.3, with FritzCard   Per Jessen
• [asterisk-users] Linksys SPA-941 Unavailable   Kim Joung-il
• [asterisk-users] Linksys SPA-941 Unavailable   Kim Joung-il
• [asterisk-users] Board configuration - specification or recommendation   Kim Joung-il
• [asterisk-users] Linksys SPA-941 Unavailable   Kim Joung-il
• [asterisk-users] Linksys SPA-941 Unavailable   Kim Joung-il
• [asterisk-users] Free T1 Card?   Michael Joyner
• [asterisk-users] SIP: "To:" header?   Michael Joyner
• [asterisk-users] IAX complaints? What are they?   Daryl G. Jurbala
• [asterisk-users] Astmanproxy Yahoo Group   Mamadou Lamine KA
• [asterisk-users] Outgoing PSTN calls , unusable voice quality   Veselin Kantsev
• [asterisk-users] Outgoing PSTN calls , unusable voice quality   Veselin Kantsev
• [asterisk-users] Phones Not Registering   Edwin Kariuki
• [asterisk-users] Phones Not Registering   Edwin Kariuki
• [asterisk-users] PRI dialout problem with some numbers...   Alejandro Kauffmann
• [asterisk-users] DST   Philipp Kempgen
• [asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered   Philipp Kempgen
• [asterisk-users] Get value from linux terminal to dialplan in Asterisk ?   Philipp Kempgen
• [asterisk-users] res_mysql versus res_odbc   Philipp Kempgen
• [asterisk-users] Snom 320 with TDM02B and echo problems   Philipp Kempgen
• [asterisk-users] "Whats New at Digium the Asterisk Company" -- Junk?   Philipp Kempgen
• [asterisk-users] Lists dead?   Philipp Kempgen
• [asterisk-users] make config update-rc.d   Philipp Kempgen
• [asterisk-users] How to enable res_config_mysql   Philipp Kempgen
• [asterisk-users] VMukti - Filesharing + video + voice supported Soft phone   Philipp Kempgen
• [asterisk-users] Snom phones, blinking lights and call pickup   Philipp Kempgen
• [asterisk-users] Restricting the manager interface to a number?   Philipp Kempgen
• [asterisk-users] Snom phones, blinking lights and call pickup   Philipp Kempgen
• [asterisk-users] Urgent question.   Philipp Kempgen
• [asterisk-users] Urgent question.   Philipp Kempgen
• [asterisk-users] Snom phones, blinking lights and call pickup   Philipp Kempgen
• [asterisk-users] Snom phones, blinking lights and call pickup   Philipp Kempgen
• [asterisk-users] Snom phones, blinking lights and call pickup   Philipp Kempgen
• [asterisk-users] cvs or svn   Philipp Kempgen
• [asterisk-users] SIP detects loop when forwarding to voicemail   Philipp Kempgen
• [asterisk-users] cvs or svn   Philipp Kempgen
• [asterisk-users] Digium TE120P versus Sangoma A101D-X   Philipp Kempgen
• [asterisk-users] To DB or not to DB?   Philipp Kempgen
• [asterisk-users] queue_log duration=NULL   Philipp Kempgen
• [asterisk-users] How to originate a call from console CLI ?   Philipp Kempgen
• [asterisk-users] IAX complaints? What are they?   Philipp Kempgen
• [asterisk-users] Mobile phone codecs ...   Steve Kennedy
• [asterisk-users] Softphone to be installed on the Mobile   Steve Kennedy
• [asterisk-users] Spam Filter News   Thomas Kenyon
• [asterisk-users] SparkLan WVPR-100 Wireless Handset.   Thomas Kenyon
• [asterisk-users] SparkLan WVPR-100 Wireless Handset.   Thomas Kenyon
• [asterisk-users] Problem with AGI Script   Mindaugas Kezys
• [asterisk-users] pip tones in Monitor or MixMonitor   Mindaugas Kezys
• [asterisk-users] pip tones in Monitor or MixMonitor   Mindaugas Kezys
• [asterisk-users] Dialing time-out   Mindaugas Kezys
• [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4   Mindaugas Kezys
• [asterisk-users] Best Prepaid Application?   Mindaugas Kezys
• [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4   Mindaugas Kezys
• [asterisk-users] Best Prepaid Application?   Mindaugas Kezys
• [asterisk-users] To DB or not to DB?   Mindaugas Kezys
• [asterisk-users] Free T1 Card?   Kristian Kielhofner
• [asterisk-users] AstLinux WebSite Problem   Kristian Kielhofner
• [asterisk-users] asterisk as non-root/best practices   Kristian Kielhofner
• [asterisk-users] Unable to lookup host in c= line,   Kristian Kielhofner
• [asterisk-users] 'Traditional' Faxing   Klaverstyn, David C
• [asterisk-users] Snom phones, blinking lights and call pickup   Philipp von Klitzing
• [asterisk-users] Snom phones, blinking lights and call pickup   Philipp von Klitzing
• [asterisk-users] Suppressing certain queue announcement voice prompts   Philipp von Klitzing
• [asterisk-users] Snom 320 with TDM02B and echo problems   Tim Koehler
• [asterisk-users] 'Traditional' Faxing   Andrew Kohlsmith
• [asterisk-users] Strange asterisk message "Remote host can't match request NOTIFY..."   Bruce Komito
• [asterisk-users] Polycom MWI's will not turn off   Bruce Komito
• [asterisk-users] SER with Asterisk intergration   Arun Kumar
• [asterisk-users] DTMF Problem   Arun Kumar
• [asterisk-users] Softphone to be installed on the Mobile   Arun Kumar
• [asterisk-users] Need help in selecting DTMF Mode   Arun Kumar
• [asterisk-users] detect asterisk pbx via sip   Mindaugas Kuprys
• [asterisk-users] ACD functionality , Skills for agents   Kyriakos
• [asterisk-users] How to integrate Asterisk with Avaya   Kyriakos
• [asterisk-users] How to integrate Asterisk with Avaya   Kyriakos
• [asterisk-users] ACD functionality , Skills for agents   Kyriakos
• [asterisk-users] ACD functionality , Skills for agents   Kyriakos
• [asterisk-users] ACD functionality , Skills for agents   Kyriakos
• [asterisk-users] ACD functionality , Skills for agents   Kyriakos
• [asterisk-users] route INVITE sip:s at sip.test.fr   Marc LEURENT
• [asterisk-users] route INVITE sip:s at sip.test.fr   Marc LEURENT
• [asterisk-users] Best Prepaid Application?   Marc LEURENT
• [asterisk-users] Best Prepaid Application?   Marc LEURENT
• [asterisk-users] Asterisk B2BUA patch useful??   Marc LEURENT
• [asterisk-users] G729 on wrong bus   Marc LEURENT
• [asterisk-users] 1 FXS module / PCI express   Marc LEURENT
• [asterisk-users] Cisco phone 7911g restarts   Paul Lacatus
• [asterisk-users] Client lost on skinny   Paul Lacatus
• [asterisk-users] Client lost on skinny   Paul Lacatus
• [asterisk-users] pipemedia   Pezhman Lali
• [asterisk-users] OT: Which SIP method to use for this specificbehaviour ?   Steve Langstaff
• [asterisk-users] OT: Which SIP method to use for thisspecificbehaviour ?   Steve Langstaff
• [asterisk-users] SLA: Handling of errors in outgoing call   Steve Langstaff
• [asterisk-users] Asterisk Recording   Lees, James (UK)
• [asterisk-users] Two B-Channel Transfer (2BCT/TBCT) Trobule on DMS100 PRI   Jacob Lefkowitz
• [asterisk-users] Recommendations for 100 Wifi SIP phone setup   Jacob Lefkowitz
• [asterisk-users] Sip to ATA?   Jacob Lefkowitz
• [asterisk-users] Error: inserting return line in dialing strings   Alejandro Lengua
• [asterisk-users] Queue Statistics reporting   Lenz
• [asterisk-users] asterisk 1.4 prereq   Lenz
• [asterisk-users] How to play Asterisk .raw file   Lenz
• [asterisk-users] Friday Conference reminder: AGI example Nov 16th at 12:30 PM EST   Lenz
• [asterisk-users] Bugtracker to use with Asterisk?   Lenz
• [asterisk-users] Queue Drops to Voicemail   Lenz
• [asterisk-users] hostname in MySQL CDR records   Tilghman Lesher
• [asterisk-users] Two PRI setup questions   Tilghman Lesher
• [asterisk-users] res_mysql versus res_odbc   Tilghman Lesher
• [asterisk-users] Route an incoming call by ANI*DNIS   Tilghman Lesher
• [asterisk-users] sip show peers in 1.4.13   Tilghman Lesher
• [asterisk-users] weird 185 secs timeout call problem   Tilghman Lesher
• [asterisk-users] How to get ten-digit number?   Tilghman Lesher
• [asterisk-users] How to get ten-digit number?   Tilghman Lesher
• [asterisk-users] asterisk ODBC dependencies   Tilghman Lesher
• [asterisk-users] Asterisk and OBDC   Tilghman Lesher
• [asterisk-users] sangoma zaptel patches   Tilghman Lesher
• [asterisk-users] function voicemailmain   Tilghman Lesher
• [asterisk-users] ODBC connection to Microsoft SQL Server   Tilghman Lesher
• [asterisk-users] Shared gsm files   Tilghman Lesher
• [asterisk-users] Asterisk Program Closes   Tilghman Lesher
• [asterisk-users] make config update-rc.d   Tilghman Lesher
• [asterisk-users] modifying a dialed exension before dialplan processing   Tilghman Lesher
• [asterisk-users] Problem with AGI Script   Tilghman Lesher
• [asterisk-users] Conference Call Dial-Out to a participant   Tilghman Lesher
• [asterisk-users] Asterisk Sound File   Tilghman Lesher
• [asterisk-users] asterisk manager and perl   Tilghman Lesher
• [asterisk-users] Pass CallerID when call forwards to PSTN?   Tilghman Lesher
• [asterisk-users] Route an incoming call by ANI*DNIS   Tilghman Lesher
• [asterisk-users] Realtime extensions configuration - calling user filtering   Tilghman Lesher
• [asterisk-users] Problem installing Asterisk   Tilghman Lesher
• [asterisk-users] spandsp as T.38 termination?   Tilghman Lesher
• [asterisk-users] Digium and Asterisk   Tilghman Lesher
• [asterisk-users] Problem installing Asterisk   Tilghman Lesher
• [asterisk-users] Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.   Tilghman Lesher
• [asterisk-users] OT Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.   Tilghman Lesher
• [asterisk-users] MSSQL ODBC Connections   Tilghman Lesher
• [asterisk-users] passing DTMF upon call answering   Tilghman Lesher
• [asterisk-users] MSSQL ODBC Connections   Tilghman Lesher
• [asterisk-users] Digium and Asterisk   Tilghman Lesher
• [asterisk-users] Digium and Asterisk   Tilghman Lesher
• [asterisk-users] Correct syntax for IF()?   Tilghman Lesher
• [asterisk-users] Multiple Return Values from func_odbc   Tilghman Lesher
• [asterisk-users] Multiple Return Values from func_odbc   Tilghman Lesher
• [asterisk-users] Adding new recorded phrases to the release   Tilghman Lesher
• [asterisk-users] Copy or Make + Make Install   Tilghman Lesher
• [asterisk-users] Using existing extensions.conf macros, and co-habitation   Tilghman Lesher
• [asterisk-users] issues with downloads.digium.com   Atis Lezdins
• [asterisk-users] ztdummy and BackGround   Atis Lezdins
• [asterisk-users] Meetme - how to protect the conference?   Atis Lezdins
• [asterisk-users] CDR on channel not posted   Atis Lezdins
• [asterisk-users] PHP - Queues - etc.   Atis Lezdins
• [asterisk-users] No sound from playback and voicemail   Atis Lezdins
• [asterisk-users] Stress-Testing Asterisk   Atis Lezdins
• [asterisk-users] How to enable res_config_mysql   Atis Lezdins
• [asterisk-users] blind transfer dumping calls   Atis Lezdins
• [asterisk-users] Asterisk Sound File   Atis Lezdins
• [asterisk-users] Asterisk Sound File   Atis Lezdins
• [asterisk-users] blind transfer dumping calls   Atis Lezdins
• [asterisk-users] Route an incoming call by ANI*DNIS   Atis Lezdins
• [asterisk-users] store 2 separate records in cdr when a call is transferd   Atis Lezdins
• [asterisk-users] Softphone to be installed on the Mobile   Atis Lezdins
• [asterisk-users] Check if SIP is avaible to dial   Atis Lezdins
• [asterisk-users] Check if SIP is avaible to dial   Atis Lezdins
• [asterisk-users] Check if SIP is avaible to dial   Atis Lezdins
• [asterisk-users] Finding the status of an extension   Atis Lezdins
• [asterisk-users] test   Atis Lezdins
• [asterisk-users] G729/MOH Quality   Atis Lezdins
• [asterisk-users] Snom 320 with TDM02B and echo problems   Michael J. Liberatore
• [asterisk-users] Kernel Native PCIE Network Cards?   Michael J. Liberatore
• [asterisk-users] Kernel Native PCIE Network Cards?   Michael J. Liberatore
• [asterisk-users] p2p t1 with sangoma hw   Michael J. Liberatore
• [asterisk-users] p2p t1 with sangoma hw   Michael J. Liberatore
• [asterisk-users] p2p t1 with sangoma hw   Michael J. Liberatore
• [asterisk-users] p2p t1 with sangoma hw   Michael J. Liberatore
• [asterisk-users] Digium and Asterisk   Michael J. Liberatore
• [asterisk-users] Annoying PRI Channels Restarting Message   Michael J. Liberatore
• [asterisk-users] Annoying PRI Channels Restarting Message   Michael J. Liberatore
• [asterisk-users] Annoying PRI Channels Restarting Message   Michael J. Liberatore
• [asterisk-users] Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.   Michael J. Liberatore
• [asterisk-users] Annoying PRI Channels Restarting Message   Michael J. Liberatore
• [asterisk-users] Digium and Asterisk   Michael J. Liberatore
• [asterisk-users] Annoying PRI Channels Restarting Message   Michael J. Liberatore
• [asterisk-users] Recommendations for 100 Wifi SIP phone setup   Michael J. Liberatore
• [asterisk-users] Nortel digital FXO channel bank? Exists?   Rupert Utteridge - Digital Techniques (Asia) Limited
• [asterisk-users] Mystery phone!   Peter Lindquist
• [asterisk-users] 7960 Queue Issue   Robert Lister
• [asterisk-users] How to delete voice mail messages?   Robert Lister
• [asterisk-users] Pickup Command not working   Robert Lister
• [asterisk-users] Changing audio message to text message   Robert Lister
• [asterisk-users] Pass CallerID when call forwards to PSTN?   Robert Lister
• [asterisk-users] Odd bug in Siemens C460IP ?   Robert Lister
• [asterisk-users] Odd bug in Siemens C460IP ?   Robert Lister
• [asterisk-users] Druid   Alan Lord
• [asterisk-users] Druid   Alan Lord
• [asterisk-users] Arbitrary limit on length of email address?   Alan Lord
• [asterisk-users] Arbitrary limit on length of email address?   Alan Lord
• [asterisk-users] Selecting OSLEC for zaptel-1.4.6   Alan Lord
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Alan Lord
• [asterisk-users] Wanted: tutorial on troubleshooting SIP issues   Alan Lord
• [asterisk-users] Polycom Speakerphone   Alan Lord
• [asterisk-users] asterisk as non-root/best practices   Alan Lord
• [asterisk-users] Softphone to be installed on the Mobile   Alan Lord
• [asterisk-users] asterisk as non-root/best practices   Alan Lord
• [asterisk-users] asterisk as non-root/best practices   Alan Lord
• [asterisk-users] Digium and Asterisk   Alan Lord
• [asterisk-users] Digium and Asterisk   Alan Lord
• [asterisk-users] Echo Cancellers. WAS Digium and Asterisk   Alan Lord
• [asterisk-users] To DB or not to DB?   Alan Lord
• [asterisk-users] To DB or not to DB?   Alan Lord
• [asterisk-users] asterisk as non-root/best practices   Alan Lord
• [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)   Luki
• [asterisk-users] What do you do to keep asterisk alive?   Luki
• [asterisk-users] RTP traffic not being forwarded   Luki
• [asterisk-users] Two PRI setup questions   Lutgring, Sam
• [asterisk-users] Pickup Command not working   Lutgring, Sam
• [asterisk-users] Pickup Command not working   Lutgring, Sam
• [asterisk-users] Pickup Command not working   Lutgring, Sam
• [asterisk-users] Remote Office, Centrally Shared Voicemail   Lutgring, Sam
• [asterisk-users] AEL2 and Callbacks   Richard Lyman
• [asterisk-users] OT: Re: How to pay for libpri development   Richard Lyman
• [asterisk-users] reload command   Julian Lyndon-Smith
• [asterisk-users] Call Failed   Doug Lytle
• [asterisk-users] Two PRI setup questions   Doug Lytle
• [asterisk-users] sip show peers in 1.4.13   Doug Lytle
• [asterisk-users] Restart when convenient   Doug Lytle
• [asterisk-users] Restart when convenient   Doug Lytle
• [asterisk-users] MeetMe CPU resources   Doug Lytle
• [asterisk-users] If caller id is null set to a specific number   Doug Lytle
• [asterisk-users] If caller id is null set to a specific number   Doug Lytle
• [asterisk-users] Kernel Native PCIE Network Cards?   Doug Lytle
• [asterisk-users] 'Traditional' Faxing   Doug Lytle
• [asterisk-users] Polycom Speakerphone   Doug Lytle
• [asterisk-users] Default mohmp3 : free of rights ?   Doug Lytle
• [asterisk-users] VoiceMail hangup   Doug Lytle
• [asterisk-users] Asterisk ignoring manager events when busy   Doug Lytle
• [asterisk-users] Reporting bugs   Doug Lytle
• [asterisk-users] [Record() function] Script stops if user doesn't hit # after msg   Doug Lytle
• [asterisk-users] Digium E1 and Digium TDM400P (2xFXO) Help!   Doug Lytle
• [asterisk-users] Asterisk version survey   Doug Lytle
• [asterisk-users] Sangoma Question   Doug Lytle
• [asterisk-users] To DB or not to DB?   Doug Lytle
• [asterisk-users] Bad audio quality in 1.4-SVN when encoding to alaw/ulaw   Doug Lytle
• [asterisk-users] Free T1 Card?   Guillermo Salas M.
• [asterisk-users] roundrobin and rrmemory with pre-defined agent order   Julian J. M.
• [asterisk-users] No ACK on 200 OK   Frederico Madeira
• [asterisk-users] Queue Drops to Voicemail   Gregory Malsack
• [asterisk-users] Sangoma Question   Jeremy Mann
• [asterisk-users] Sangoma Question   Jeremy Mann
• [asterisk-users] DeStar finalist in Les Trophées du Libre 2007   Juan Manuel
• [asterisk-users] California based PSTN connections   Adrian Marsh
• [asterisk-users] SIP port 5060 closed - how do I open it?   Adrian Marsh
• [asterisk-users] SIP port 5060 closed - how do I open it?   Adrian Marsh
• [asterisk-users] Intercom with Snom phones   Zaheer Master
• [asterisk-users] AsteriskNOW - how to open SIP ports?   Zaheer K. Master
• [asterisk-users] Install Scripts for CentOS 4   Zaheer K. Master
• [asterisk-users] Install Scripts for CentOS 4   Zaheer K. Master
• [asterisk-users] Problem receiving audio with Asterisk 1.4.5 with SIP trunk   Zaheer K. Master
• [asterisk-users] Problem receiving audio with Asterisk 1.4.5 withSIP trunk   Zaheer K. Master
• [asterisk-users] SIP port 5060 closed - how do I open it?   Zaheer K. Master
• [asterisk-users] SIP port 5060 closed - how do I open it?   Zaheer K. Master
• [asterisk-users] SIP port 5060 closed - how do I open it?   Zaheer K. Master
• [asterisk-users] SIP port 5060 closed - how do I open it?   Zaheer K. Master
• [asterisk-users] SIP port 5060 closed - how do I open it?   Zaheer K. Master
• [asterisk-users] SIP port 5060 closed - how do I open it?   Zaheer K. Master
• [asterisk-users] SIP port 5060 closed - how do I open it?   Zaheer K. Master
• [asterisk-users] SIP port 5060 closed - how do I open it?   Zaheer K. Master
• [asterisk-users] Problem with AGI Script   Matt
• [asterisk-users] Problem with AGI Script   Matt
• [asterisk-users] IVR Tree Best Practices   Matt
• [asterisk-users] Problem with AGI Script   Matt
• [asterisk-users] Problem with AGI Script   Matt
• [asterisk-users] Problem installing Asterisk   Matt
• [asterisk-users] Problem with AGI Script   Matt
• [asterisk-users] Problem installing Asterisk   Matt
• [asterisk-users] Asterisk behind a PIX firewall?   Matt
• [asterisk-users] Asterisk behind a PIX firewall?   Matt
• [asterisk-users] Asterisk behind a PIX firewall?   Matt
• [asterisk-users] Asterisk behind a PIX firewall?   Matt
• [asterisk-users] Asterisk behind a PIX firewall?   Matt
• [asterisk-users] Asterisk behind a PIX firewall?   Matt
• [asterisk-users] Asterisk behind a PIX firewall?   Matt
• [asterisk-users] Asterisk behind a PIX firewall?   Matt
• [asterisk-users] Asterisk behind a PIX firewall?   Matt
• [asterisk-users] Asterisk behind a PIX firewall?   Matt
• [asterisk-users] Asterisk behind a PIX firewall?   Matt
• [asterisk-users] Asterisk behind a PIX firewall?   Matt
• [asterisk-users] Asterisk behind a PIX firewall?   Matt
• [asterisk-users] 'h' extension on call-out   Artifex Maximus
• [asterisk-users] 'h' extension on call-out   Artifex Maximus
• [asterisk-users] 'h' extension on call-out   Artifex Maximus
• [asterisk-users] 'h' extension on call-out   Artifex Maximus
• [asterisk-users] Shared line appearance phones?   Ron McCarthy
• [asterisk-users] Shared line appearance phones?   Ron McCarthy
• [asterisk-users] OT: Best firmware for Linksys Router thatis "SIP AWARE"   Ron McCarthy
• [asterisk-users] Shared line appearance phones?   Ron McCarthy
• [asterisk-users] [asterisk-dev] trunk working under windows!   Max McGraw
• [asterisk-users] asterisk ODBC dependencies   Robert McNaught
• [asterisk-users] ODBC connection to Microsoft SQL Server   Robert McNaught
• [asterisk-users] asterisk as non-root/best practices   Robert McNaught
• [asterisk-users] dialplan design - unknown extension length   Robert McNaught
• [asterisk-users] dialplan design - unknown extension length   Robert McNaught
• [asterisk-users] asterisk as non-root/best practices   Robert McNaught
• [asterisk-users] sip proxy failover   Robert McNaught
• [asterisk-users] MSSQL ODBC Connections   Robert McNaught
• [asterisk-users] MSSQL ODBC Connections   Robert McNaught
• [asterisk-users] passing DTMF upon call answering   Robert McNaught
• [asterisk-users] To DB or not to DB?   Robert McNaught
• [asterisk-users] asterisk as non-root/best practices   Robert McNaught
• [asterisk-users] asterisk as non-root/best practices   Robert McNaught
• [asterisk-users] asterisk as non-root/best practices   Robert McNaught
• [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)   Antoine Megalla
• [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)   Antoine Megalla
• [asterisk-users] Call Forward on SIP unreachable (network failure)   Antoine Megalla
• [asterisk-users] Get value from linux terminal to dialplan in Asterisk ?   Arpit Mehta
• [asterisk-users] Get value from linux terminal to dialplan in Asterisk ?   Arpit Mehta
• [asterisk-users] use dial plan passed arg value in C agi code   Arpit Mehta
• [asterisk-users] use dial plan passed arg value in C agi code   Arpit Mehta
• [asterisk-users] Get IP address of an incoming or outgoing SIP call   Arpit Mehta
• [asterisk-users] Get IP address of an incoming or outgoing SIP call   Arpit Mehta
• [asterisk-users] REFER mesage extraction using SIP_HEADER   Arpit Mehta
• [asterisk-users] Autodialing   Bhrugu Mehta
• [asterisk-users] Need Reference sites   Bhrugu Mehta
• [asterisk-users] Queue   Bhrugu Mehta
• [asterisk-users] MSSQL ODBC Connections   Bhrugu Mehta
• [asterisk-users] MSSQL ODBC Connections   Bhrugu Mehta
• [asterisk-users] 7960 Queue Issue   Eric Merkel
• [asterisk-users] asterisk-stat problem   Anthony Messina
• [asterisk-users] hostname in MySQL CDR records   Anthony Messina
• [asterisk-users] hostname in MySQL CDR records   Anthony Messina
• [asterisk-users] Asterisk version survey   Anthony Messina
• [asterisk-users] Registration problem: UA -> SER -> Asterisk   Giovanni Miano
• [asterisk-users] Registration problem: UA -> SER -> Asterisk   Giovanni Miano
• [asterisk-users] Suppressing certain queue announcement voice prompts   Mark Michelson
• [asterisk-users] Suppressing certain queue announcement voice prompts   Mark Michelson
• [asterisk-users] Calling with hidden callerid   Mike
• [asterisk-users] Calling with hidden callerid   Mike
• [asterisk-users] Queue with cell phones   Mike
• [asterisk-users] Do While loop   Mike
• [asterisk-users] e911   Dave Miller
• [asterisk-users] Polycom Speakerphone   John Millican
• [asterisk-users] Polycom Speakerphone   John Millican
• [asterisk-users] Sip 1.4.x DTMF detection not working   John Millican
• [asterisk-users] flooded by "Maximum trunk data space exceeded" messages   Louis-David Mitterrand
• [asterisk-users] Correct syntax for IF()?   Adam Moffett
• [asterisk-users] Newb Question   Adam Moffett
• [asterisk-users] Which files to be copied   Mojo with Horan & Company, LLC
• [asterisk-users] dialplan design - unknown extension length   Mojo with Horan & Company, LLC
• [asterisk-users] dialplan design - unknown extension length   Mojo with Horan & Company, LLC
• [asterisk-users] Caller ID Question   Mojo with Horan & Company, LLC
• [asterisk-users] Asterisk API Manager   Mojo with Horan & Company, LLC
• [asterisk-users] Asterisk API Manager   Mojo with Horan & Company, LLC
• [asterisk-users] Copy or Make + Make Install   Mojo with Horan & Company, LLC
• [asterisk-users] Do While loop   Mojo with Horan & Company, LLC
• [asterisk-users] Copy or Make + Make Install   Mojo with Horan & Company, LLC
• [asterisk-users] sidetone   Mojo with Horan & Company, LLC
• [asterisk-users] "Whats New at Digium the Asterisk Company" -- Junk?   Jesse Molina
• [asterisk-users] p2p t1 with sangoma hw   Jesse Molina
• [asterisk-users] Trouble with asterisk-users mailman   Jesse Molina
• [asterisk-users] Trouble with asterisk-users mailman   Jesse Molina
• [asterisk-users] Trouble with asterisk-users mailman   Jesse Molina
• [asterisk-users] test   Jesse Molina
• [asterisk-users] Off-Topic: Avaya   Jesse Molina
• [asterisk-users] Problem with CDR userfield not being set   James Moore
• [asterisk-users] Problem with CDR userfield not being set   James Moore
• [asterisk-users] Pickup Command not working   Lacy Moore
• [asterisk-users] Music on Hold Problem w/ Transfers   Lacy Moore
• [asterisk-users] Music on Hold Problem w/ Transfers   Lacy Moore
• [asterisk-users] Shared line appearance phones?   Lacy Moore
• [asterisk-users] TE210P Vs TE220P difference