May 2010 Archives by author

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Starting: Sat May 1 02:38:34 CDT 2010
Ending: Mon May 31 21:29:23 CDT 2010
Messages: 1064

• [asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server   William Stillwell (Lists)
• [asterisk-users] Need fax solution for 1.4.xx   William Stillwell (Lists)
• [asterisk-users] Need fax solution for 1.4.xx   William Stillwell (Lists)
• [asterisk-users] Need fax solution for 1.4.xx   William Stillwell (Lists)
• [asterisk-users] Need fax solution for 1.4.xx   William Stillwell (Lists)
• [asterisk-users] Need fax solution for 1.4.xx (Resolution)   William Stillwell (Lists)
• [asterisk-users] Need fax solution for 1.4.xx (Resolution)   William Stillwell (Lists)
• [asterisk-users] Softphones on thin clients...   William Stillwell (Lists)
• [asterisk-users] Music on Hold   William Stillwell (Lists)
• [asterisk-users] Interesting email project.   adamk at 3a.hu
• [asterisk-users] client-server encryption   adamk at 3a.hu
• [asterisk-users] What is billsec in CDR?   adamk at 3a.hu
• [asterisk-users] PRI down due to chan_zap.c: No more room in scheduler....Got SABME and Sending Unnumbered Acknowledgement...Any thoughts?   Gopalakrishnan A.N
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Gopalakrishnan A.N
• [asterisk-users] Sending fake auth rejection for user   Gopalakrishnan A.N
• [asterisk-users] About Sangoma cards and Asterisk integration with other PBX   Gopalakrishnan A.N
• [asterisk-users] Asterisk and Patton   A.Santoro
• [asterisk-users] Aastra i740 and Asterisk   Alyed
• [asterisk-users] Pattern matching - how to ignore numbers after 10 digits   Alyed
• [asterisk-users] working example of t38 fax w/ 1.6.2?   Miguel Amez
• [asterisk-users] sending T.38 fax negotiation problem   Miguel Amez
• [asterisk-users] working example of t38 fax w/ 1.6.2?   Miguel Amez
• [asterisk-users] sending T.38 fax negotiation problem   Miguel Amez
• [asterisk-users] VoIP Termination in Japan   Miguel Amez
• [asterisk-users] T38 trunk configuration for relay appears to affect default trunks for voip   Miguel Amez
• [asterisk-users] Little t38 bug?   Miguel Amez
• [asterisk-users] Little t38 bug?   Miguel Amez
• [asterisk-users] Conflict between jumper settings and dmesg with OctoBRI   Andre
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Andres
• [asterisk-users] Installing sounds   Andres
• [asterisk-users] voice recognition suggestion   Rilawich Ango
• [asterisk-users] TTS for asterisk   Rilawich Ango
• [asterisk-users] TTS for asterisk   Rilawich Ango
• [asterisk-users] TTS for asterisk   Rilawich Ango
• [asterisk-users] SPA-932 BLF Stays solid red for single extension   Steve Anness
• [asterisk-users] SPA-932 BLF Stays solid red for singleextension   Steve Anness
• [asterisk-users] Records sets and ODBC   Lee Archer
• [asterisk-users] Continue dialplan is source channel hangs up   Lee Archer
• [asterisk-users] Continue dialplan is source channel hangs up   Lee Archer
• [asterisk-users] Records sets and ODBC   Lee Archer
• [asterisk-users] Continue dialplan is source channel hangs up   Lee Archer
• [asterisk-users] Have a macro update a channel variable   Lee Archer
• [asterisk-users] Adding a context from the console   Lee Archer
• [asterisk-users] Adding a context from the console   Lee Archer
• [asterisk-users] Adding a context from the console   Lee Archer
• [asterisk-users] Adding a context from the console   Lee Archer
• [asterisk-users] Help with IP Routing   Lee Archer
• [asterisk-users] Adding a context from the console   Lee Archer
• [asterisk-users] Conflict between jumper settings and dmesg with?OctoBRI   Ron Arts
• [asterisk-users] Implementing Asterisk on a WAN or an Intranet   Goke M Aruna
• [asterisk-users] Play MusicOnHold and continue with dialplan   Asterisk
• [asterisk-users] Play MusicOnHold and continue with dialplan   Asterisk
• [asterisk-users] Play MusicOnHold and continue with dialplan   Asterisk
• [asterisk-users] DID's for Chatham, ON   Robert Augustyn
• [asterisk-users] BADTIME FOR ANSWEREDTIME   François BERGANZ
• [asterisk-users] BAD ROUND TIME FOR ANSWEREDTIME   François BERGANZ
• [asterisk-users] CANCEL Reason   François BERGANZ
• [asterisk-users] CANCEL Reason   François BERGANZ
• [asterisk-users] quality of the call at the hangup   François BERGANZ
• [asterisk-users] CANCEL Reason   François BERGANZ
• [asterisk-users] long return times from System() calls with 1.6.2.6?   David Backeberg
• [asterisk-users] Questions About Fax for Asterisk   David Backeberg
• [asterisk-users] Speech/DTMF mix?   David Backeberg
• [asterisk-users] Digits and Vestec   David Backeberg
• [asterisk-users] Digits and Vestec   David Backeberg
• [asterisk-users] Need fax solution for 1.4.xx   David Backeberg
• [asterisk-users] Problem with callerid(dnid) and queue   David Backeberg
• [asterisk-users] Need fax solution for 1.4.xx   David Backeberg
• [asterisk-users] Asterisk core dumping on SendFax with FFA   David Backeberg
• [asterisk-users] Asterisk 1.4.30 & T38   David Backeberg
• [asterisk-users] Asterisk Cluster   David Backeberg
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   David Backeberg
• [asterisk-users] [0017330] 1.6.1 and 1.6.2 + MySQL crases on ODBC Query (via func_odbc or sip realtime)   David Backeberg
• [asterisk-users] Meetmee user introduction disabled   David Backeberg
• [asterisk-users] Meetmee user introduction disabled   Theo Band
• [asterisk-users] Meetmee user introduction disabled   Theo Band
• [asterisk-users] Security tests   Daniel Bareiro
• [asterisk-users] Problem with Music on hold   Daniel Bareiro
• [asterisk-users] Callerid with DAHDI   Daniel Bareiro
• [asterisk-users] Callerid with DAHDI   Daniel Bareiro
• [asterisk-users] About Sangoma cards and Asterisk integration with other PBX   Daniel Bareiro
• [asterisk-users] About Sangoma cards and Asterisk integration with other PBX   Daniel Bareiro
• [asterisk-users] About Sangoma cards and Asterisk integration with other PBX   Daniel Bareiro
• [asterisk-users] Channel failover   Jack Bates
• [asterisk-users] Channel failover   Jack Bates
• [asterisk-users] Problem of "Playing 'pbx-transfer'"   Dovid Bender
• [asterisk-users] Getting presence working in 1.6.2   Dovid Bender
• [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration   Hristo Benev
• [asterisk-users] routing of calls   Trevor Benson
• [asterisk-users] Better AMD module   Trevor Benson
• [asterisk-users] "ring splash"   Trevor Benson
• [asterisk-users] OT: NAT in SPA922   Vineet Bhojnagarwala
• [asterisk-users] OT: NAT in SPA922   Vineet Bhojnagarwala
• [asterisk-users] Getting presence working in 1.6.2   Gareth Blades
• [asterisk-users] Getting presence working in 1.6.2   Gareth Blades
• [asterisk-users] "Contact header appears incorrect on this invite" Asterisk registering with another PBX   Gareth Blades
• [asterisk-users] "Contact header appears incorrect on this invite" Asterisk registering with another PBX   Gareth Blades
• [asterisk-users] "Contact header appears incorrect on this invite" Asterisk registering with another PBX   Gareth Blades
• [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality   Gareth Blades
• [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality   Gareth Blades
• [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality   Gareth Blades
• [asterisk-users] identify caller hangup or callee hangup?   Gareth Blades
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Gareth Blades
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Gareth Blades
• [asterisk-users] DAHDI Help (made a cardinal sin :()   Gareth Blades
• [asterisk-users] Need fax solution for 1.4.xx   Rod Boileau
• [asterisk-users] client-server encryption   Jeff Brower
• [asterisk-users] voipmonitor.org   Jeff Brower
• [asterisk-users] OK, I'm stumped   Jeff Brower
• [asterisk-users] OK, I'm stumped   Jeff Brower
• [asterisk-users] Asterisk Cluster   Jeff Brower
• [asterisk-users] Asterisk Cluster   Jeff Brower
• [asterisk-users] ControlPlayback skip forward fails on mp3 file   Russell Brown
• [asterisk-users] Contact header gets url decoded?   Tom Browning
• [asterisk-users] Agent Privacy - chan_local   Robert Broyles
• [asterisk-users] OK, I'm stumped   Pascal Bruno
• [asterisk-users] AGI and Severe Weather Alerts   Frank Bulk
• [asterisk-users] Problem with AMI Originate   Leo Burd
• [asterisk-users] Problems with Asterisk and two Linksys SPA941   Olivier CALVANO
• [asterisk-users] How to use one single IP as origination   CDR
• [asterisk-users] Cant load chan_dahdi   Emanuele Carbone
• [asterisk-users] VoIP Termination in Japan   Justin Case
• [asterisk-users] Reloading queue members (realtime DB)   Aksel Celasun
• [asterisk-users] DAHDI astribank Channel Unavailable   Kelvin Chan
• [asterisk-users] CDR to MS-SQL via ODBC issue   Neeraj Chand
• [asterisk-users] CDR to MS-SQL via ODBC issue   Neeraj Chand
• [asterisk-users] CDR to MS-SQL via ODBC issue   Neeraj Chand
• [asterisk-users] realtime queues "membername" problem   Jean Chassoul
• [asterisk-users] Are there AMI commands to manipulate a voice mailbox?   Carlos Chavez
• [asterisk-users] Softphones on thin clients...   Carlos Chavez
• [asterisk-users] Softphones on thin clients...   Carlos Chavez
• [asterisk-users] convert zaptel to dahdi?   Carlos Chavez
• [asterisk-users] [Dahdi/system.conf] "fxsks = 1" deprecated?   Carlos Chavez
• [asterisk-users] Asterisk Cluster   Adolphe Cher-Aime
• [asterisk-users] Problem of "Playing 'pbx-transfer'"   Adolphe Cher-aime
• [asterisk-users] OK, I'm stumped   Adolphe Cher-aime
• [asterisk-users] Asterisk Cluster   Adolphe Cher-aime
• [asterisk-users] Asterisk Cluster   Adolphe Cher-aime
• [asterisk-users] Asterisk Cluster   Adolphe Cher-aime
• [asterisk-users] Asterisk Cluster   Adolphe Cher-aime
• [asterisk-users] Agents   Peter Childs
• [asterisk-users] Agents   Peter Childs
• [asterisk-users] State of JACK support i9n Asterisk   Julien Claassen
• [asterisk-users] State of JACK support i9n Asterisk   Julien Claassen
• [asterisk-users] Jack in /usr/local/ means failure for asterisk   Julien Claassen
• [asterisk-users] Jack in /usr/local/ means failure for asterisk   Julien Claassen
• [asterisk-users] Jack in /usr/local/ means failure for asterisk   Julien Claassen
• [asterisk-users] Jack in /usr/local/ means failure for asterisk   Julien Claassen
• [asterisk-users] testing my asterisk 1.6.2.8-rc1 with gtalk (and JACK) - please help   Julien Claassen
• [asterisk-users] Definie gtalk troubles over here   Julien Claassen
• [asterisk-users] Definie gtalk troubles over here   Julien Claassen
• [asterisk-users] AGI and Severe Weather Alerts   Seann Clark
• [asterisk-users] Asterisk Sip Proxies and SIP persistence   Seann Clark
• [asterisk-users] problem with trustrpid   Jesse Cloutier
• [asterisk-users] conf files vs astdb   Harel Cohen
• [asterisk-users] JNET's qozap, dahdi and PCI-E Quad   Tzafrir Cohen
• [asterisk-users] DAHDI astribank Channel Unavailable   Tzafrir Cohen
• [asterisk-users] Cant load chan_dahdi   Tzafrir Cohen
• [asterisk-users] client-server encryption   Tzafrir Cohen
• [asterisk-users] DAHDI not detecting hangup   Tzafrir Cohen
• [asterisk-users] Xorcom Astribank One call and dies   Tzafrir Cohen
• [asterisk-users] conf files vs astdb   Tzafrir Cohen
• [asterisk-users] Voicemail() app not available?   Tzafrir Cohen
• [asterisk-users] Re-compiling q931.c   Tzafrir Cohen
• [asterisk-users] Re-compiling q931.c   Tzafrir Cohen
• [asterisk-users] OK, I'm stumped   Tzafrir Cohen
• [asterisk-users] Commands to upgrade to latest Libpri? can I upgrade without touching Asterisk?   Tzafrir Cohen
• [asterisk-users] Callerid with DAHDI   Tzafrir Cohen
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Tzafrir Cohen
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Tzafrir Cohen
• [asterisk-users] Dahdi problems with kernel 2.6.32   Tzafrir Cohen
• [asterisk-users] Dahdi problems with kernel 2.6.32   Tzafrir Cohen
• [asterisk-users] convert zaptel to dahdi?   Tzafrir Cohen
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Tzafrir Cohen
• [asterisk-users] routing of calls   Tzafrir Cohen
• [asterisk-users] BRI card(B800P) doesn's work with DAHDI(wcb4xxp) in NT mode   Tzafrir Cohen
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Tzafrir Cohen
• [asterisk-users] Conflict between jumper settings and dmesg with?OctoBRI   Tzafrir Cohen
• [asterisk-users] DAHDI Help (made a cardinal sin :()   Tzafrir Cohen
• [asterisk-users] Conflict between jumper settings and dmesg with?OctoBRI   Tzafrir Cohen
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Tzafrir Cohen
• [asterisk-users] Microsoft Response Point Voip server discontinued   Dean Collins
• [asterisk-users] AMR codec for Asterisk 1.6.1.X   Andrea Cristofanini
• [asterisk-users] AMR codec for Asterisk 1.6.1.X   Andrea Cristofanini
• [asterisk-users] Is it possible to get video working on h323 calls?   Kevin Cross
• [asterisk-users] Converting video files into .h263   Kevin Cross
• [asterisk-users] One way audio problem, a=sendonly and a re-invite   David Cunningham
• [asterisk-users] What does Asterisk give to reject a re-invite?   David Cunningham
• [asterisk-users] What does Asterisk give to reject a re-invite?   David Cunningham
• [asterisk-users] What does Asterisk give to reject a re-invite?   David Cunningham
• [asterisk-users] Cause and cure for "Exceptionally long voice queue length queuing to Local"?   David Cunningham
• [asterisk-users] Cause and cure for "Exceptionally long voice queuelength queuing to Local"?   David Cunningham
• [asterisk-users] Cause and cure for "Exceptionally long voice queuelength queuing to Local"?   David Cunningham
• [asterisk-users] Cause and cure for "Exceptionally long voice queue length queuing to Local"?   David Cunningham
• [asterisk-users] Using unix socket to connect with database   Deepesh D
• [asterisk-users] Using unix socket to connect with database   Deepesh D
• [asterisk-users] Using unix socket to connect with database   Deepesh D
• [asterisk-users] Getting 'username' of sip peer   Deepesh D
• [asterisk-users] Getting 'username' of sip peer   Deepesh D
• [asterisk-users] Getting 'username' of sip peer   Deepesh D
• [asterisk-users] Stress Test new system   Klaus Darilion
• [asterisk-users] Asterisk Bible?   Klaus Darilion
• [asterisk-users] Possible bug in chan_sip:add_sdp   Klaus Darilion
• [asterisk-users] identify caller hangup or callee hangup?   Klaus Darilion
• [asterisk-users] Getting presence working in 1.6.2   Matt Darnell
• [asterisk-users] AsteriskNow   Dave
• [asterisk-users] "ring splash"   Brent Davidson
• [asterisk-users] "ring splash"   Brent Davidson
• [asterisk-users] SIP and codec negotiation   Steve Davies
• [asterisk-users] OT: Windows TAPI command-line driver   Steve Davies
• [asterisk-users] "bad magic number" log messages   Alec Davis
• [asterisk-users] "bad magic number" log messages   Alec Davis
• [asterisk-users] DAHDI and ESXi   Alec Davis
• [asterisk-users] Execute AGI, Then Continue   Randall Degges
• [asterisk-users] Asterisk Bible?   Tim Densmore
• [asterisk-users] res_config_mysql - maximum field length for appdata   Sebastian Denz
• [asterisk-users] Creating a HTTP Request on missed call?   Sebastian Denz
• [asterisk-users] Multiple SIP lines.   Jim Dickenson
• [asterisk-users] More clarification on outbound sip channels.   Jim Dickenson
• [asterisk-users] Are there AMI commands to manipulate a voice mailbox?   Jim Dickenson
• [asterisk-users] play a sound file directly to a caller channel   Jim Dickenson
• [asterisk-users] OK, I'm stumped   Jim Dickenson
• [asterisk-users] Agents   Jim Dickenson
• [asterisk-users] DTMF Input from the User   Jim Dickenson
• [asterisk-users] Sip phone does not call   Jim Dickenson
• [asterisk-users] run extensions after call moved to queue and answered by member   Jim Dickenson
• [asterisk-users] Using Sangoma Call Progress Analysis behind NAT router   Jim Dickenson
• [asterisk-users] Problem with callerid(dnid) and queue   Carlo Dimaggio
• [asterisk-users] Problem with callerid(dnid) and queue   Carlo Dimaggio
• [asterisk-users] Problem with callerid(dnid) and queue   Carlo Dimaggio
• [asterisk-users] Asterisk core dumping on SendFax with FFA   Ben Dinnerville
• [asterisk-users] Asterisk core dumping on SendFax with FFA   Ben Dinnerville
• [asterisk-users] Asterisk core dumping on SendFax with FFA   Ben Dinnerville
• [asterisk-users] Asterisk Cluster   Michelle Dupuis
• [asterisk-users] Can't load ooh323 on Centos x86_64: capabilities failure   Michelle Dupuis
• [asterisk-users] How to use one single IP as origination   Michelle Dupuis
• [asterisk-users] AGI <==> DeadAGI   Steve Edwards
• [asterisk-users] Calling a RESTful Web service from Dialplan????   Steve Edwards
• [asterisk-users] BADTIME FOR ANSWEREDTIME   Steve Edwards
• [asterisk-users] text   Steve Edwards
• [asterisk-users] Manipulating the Blacklist database   Steve Edwards
• [asterisk-users] text   Steve Edwards
• [asterisk-users] AGI and Severe Weather Alerts   Steve Edwards
• [asterisk-users] Asterisk Bible?   Steve Edwards
• [asterisk-users] include sip configuration from another file in sip.conf   Steve Edwards
• [asterisk-users] Error at start of asterisk with cdr_addon_mysql.o   Steve Edwards
• [asterisk-users] Do you think my server is being attacked?   Steve Edwards
• [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality   Steve Edwards
• [asterisk-users] OK, I'm stumped   Steve Edwards
• [asterisk-users] quick question on conf bridge   Steve Edwards
• [asterisk-users] TTS for asterisk   Steve Edwards
• [asterisk-users] Pattern matching - how to ignore numbers after 10 digits   Steve Edwards
• [asterisk-users] Best way to limit outgoing calls per trunk   Steve Edwards
• [asterisk-users] Check if extension loaded over AMI   Lenz Emilitri
• [asterisk-users] Problem with callerid(dnid) and queue   Lenz Emilitri
• [asterisk-users] Agents   Lenz Emilitri
• [asterisk-users] call-waiting   C F
• [asterisk-users] Digium TE121P + DAHDI   Sascha Ferley
• [asterisk-users] Best way to limit outgoing calls per trunk   Helius Ferreira
• [asterisk-users] 1.6.2.7 SIP realtime problem   Bruce Ferrell
• [asterisk-users] 1.6.2.7 SIP realtime problem   Bruce Ferrell
• [asterisk-users] OK, I'm stumped   Bruce Ferrell
• [asterisk-users] Productivity Suite on Polycom IP7000   Karl Fife
• [asterisk-users] Productivity Suite on Polycom IP7000   Karl Fife
• [asterisk-users] Asterisk Bible?   Karl Fife
• [asterisk-users] NPA NXX Database   Cary Fitch
• [asterisk-users] "ring splash"   Cary Fitch
• [asterisk-users] sending T.38 fax negotiation problem   Kevin P. Fleming
• [asterisk-users] sending T.38 fax negotiation problem   Kevin P. Fleming
• [asterisk-users] Asterisk 1.6.2.7 Now Available   Kevin P. Fleming
• [asterisk-users] T.38 Fax With Flowroute SIP Provider   Kevin P. Fleming
• [asterisk-users] Video in Skype for Asterisk   Kevin P. Fleming
• [asterisk-users] voipmonitor.org   Kevin P. Fleming
• [asterisk-users] Asterisk core dumping on SendFax with FFA   Kevin P. Fleming
• [asterisk-users] Need fax solution for 1.4.xx   Kevin P. Fleming
• [asterisk-users] What does Asterisk give to reject a re-invite?   Kevin P. Fleming
• [asterisk-users] What does Asterisk give to reject a re-invite?   Kevin P. Fleming
• [asterisk-users] Using unix socket to connect with database   Kevin P. Fleming
• [asterisk-users] Using unix socket to connect with database   Kevin P. Fleming
• [asterisk-users] Using unix socket to connect with database   Kevin P. Fleming
• [asterisk-users] sip and SSL   Kevin P. Fleming
• [asterisk-users] Little t38 bug?   Kevin P. Fleming
• [asterisk-users] "ring splash"   Kevin P. Fleming
• [asterisk-users] Switchvox vs Asterisk codebase   Kevin P. Fleming
• [asterisk-users] 180 with SDP   Dovey Forman
• [asterisk-users] What is billsec in CDR?   BERGANZ Francois
• [asterisk-users] Voicemail() app not available?   Andrew Furey
• [asterisk-users] Voicemail() app not available?   Andrew Furey
• [asterisk-users] Are there AMI commands to manipulate a voice mailbox?   Andrew Furey
• [asterisk-users] How to have Asterisk respond from the IP address used for registration   Andrew Furey
• [asterisk-users] Aastra SIP phone regisration problems   Jose Flores Galicia
• [asterisk-users] OK, I'm stumped   Jose Flores Galicia
• [asterisk-users] OK, I'm stumped   Jose Flores Galicia
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Jose Flores Galicia
• [asterisk-users] Interesting email project.   Jian Gao
• [asterisk-users] What is billsec in CDR?   Jian Gao
• [asterisk-users] What is billsec in CDR?   Jian Gao
• [asterisk-users] What is billsec in CDR?   Jian Gao
• [asterisk-users] SIP SRV Registration problem   Jian Gao
• [asterisk-users] SIP SRV Registration problem   Jian Gao
• [asterisk-users] Connect mobile to asterisk   Tiago Geada
• [asterisk-users] shoretel pbx   Jerry Geis
• [asterisk-users] shoretel pbx   Jerry Geis
• [asterisk-users] shoretel pbx - getting PRI call status not the SIP PBX call status.   Jerry Geis
• [asterisk-users] quick question on conf bridge   Jerry Geis
• [asterisk-users] sip and SSL   Jerry Geis
• [asterisk-users] nortel meridian question   Jerry Geis
• [asterisk-users] B400P card crashes conncection   Peter Gelencser
• [asterisk-users] mISDN compiling error   Peter Gelencser
• [asterisk-users] Connect mobile to asterisk   Gilles
• [asterisk-users] Connect mobile to asterisk   Gilles
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Gilles
• [asterisk-users] test message   GlenM
• [asterisk-users] Caller ID questions   GlenM
• [asterisk-users] Caller ID questions   GlenM
• [asterisk-users] Unable to hear voice when called from PSTN phone line   Supratik Goswami
• [asterisk-users] Softphones on thin clients...   Michael Graves
• [asterisk-users] IAX2 - providers discontinuing support   Joe Greco
• [asterisk-users] Dahdi PRI T1 Setup for TE210P   Al Grims
• [asterisk-users] VoIP over virtualized VPN   Andrew Hakman
• [asterisk-users] How to use one single IP as origination   Peter den Hartog
• [asterisk-users] About option U in Dial Ast version 1.6.2   Vardan Harutyunyan
• [asterisk-users] CANCEL Reason   Vardan Harutyunyan
• [asterisk-users] Best way to limit outgoing calls per trunk   Vardan Harutyunyan
• [asterisk-users] Best way to limit outgoing calls per trunk   Vardan Harutyunyan
• [asterisk-users] Best way to limit outgoing calls per trunk   Vardan Harutyunyan
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Gordon Henderson
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Gordon Henderson
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Gordon Henderson
• [asterisk-users] Caller ID questions   Gordon Henderson
• [asterisk-users] N900 video with Asterisk?   Gordon Henderson
• [asterisk-users] How to have Asterisk respond from the IP address used for registration   Gordon Henderson
• [asterisk-users] DAHDI Help (made a cardinal sin :()   Gordon Henderson
• [asterisk-users] Getting "ghost" transfer or music on hold   Fabiano Carlos Heringer
• [asterisk-users] Getting "ghost" transfer or music on hold   Fabiano Carlos Heringer
• [asterisk-users] Which issue is keeping you from updrading to1.6.2 ?   Rob Hillis
• [asterisk-users] Normalizing called numbers   Johann Hoehn
• [asterisk-users] sending T.38 fax negotiation problem   Lee Howard
• [asterisk-users] Channel failover   Steve Howes
• [asterisk-users] OT: NAT in SPA922   Steve Howes
• [asterisk-users] OT: NAT in SPA922   Steve Howes
• [asterisk-users] Softphones on thin clients...   Steve Howes
• [asterisk-users] DAHDI not detecting hangup   Brian C. Huffman
• [asterisk-users] DAHDI not detecting hangup   Brian C. Huffman
• [asterisk-users] Pass MWI from analog line through DAHDI?   Brian C. Huffman
• [asterisk-users] Calling a RESTful Web service from Dialplan????   DHAVAL INDRODIYA
• [asterisk-users] [ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.   DHAVAL INDRODIYA
• [asterisk-users] [ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.   DHAVAL INDRODIYA
• [asterisk-users] [Asterisk-User] Asterisk Video support   DHAVAL INDRODIYA
• [asterisk-users] AskoziaPBX 2.0 Released   Michael Iedema
• [asterisk-users] Getting presence working in 1.6.2   Ira
• [asterisk-users] Asterisk stopping for no reason   Motiejus Jakštys
• [asterisk-users] Calling a RESTful Web service from Dialplan????   Motiejus Jakštys
• [asterisk-users] Reading the CDR   Motiejus Jakštys
• [asterisk-users] Getting calee audio in Asterisk (real time)   Motiejus Jakštys
• [asterisk-users] Getting calee audio in Asterisk (real time)   Motiejus Jakštys
• [asterisk-users] Make the call finish after executing Dial(G())   Motiejus Jakštys
• [asterisk-users] Make the call finish after executing Dial(G())   Motiejus Jakštys
• [asterisk-users] Getting calee audio in Asterisk (real time)   Motiejus Jakštys
• [asterisk-users] Simulating a commercial SIP provider   Motiejus Jakštys
• [asterisk-users] Simulating a commercial SIP provider   Motiejus Jakštys
• [asterisk-users] Creating a HTTP Request on missed call?   Motiejus Jakštys
• [asterisk-users] Additional CDR values   Motiejus Jakštys
• [asterisk-users] Are there AMI commands to manipulate a voice mailbox?   Motiejus Jakštys
• [asterisk-users] new way to capture audio streams in calls   Motiejus Jakštys
• [asterisk-users] [ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.   Motiejus Jakštys
• [asterisk-users] [ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.   Motiejus Jakštys
• [asterisk-users] [ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.   Motiejus Jakštys
• [asterisk-users] Early injecting Jack between call parties   Motiejus Jakštys
• [asterisk-users] Connecting 1-2 GSM ports to asterisk?   Motiejus Jakštys
• [asterisk-users] Installing sounds   Motiejus Jakštys
• [asterisk-users] State of JACK support i9n Asterisk   Motiejus Jakštys
• [asterisk-users] Jack in /usr/local/ means failure for asterisk   Motiejus Jakštys
• [asterisk-users] Jack in /usr/local/ means failure for asterisk   Motiejus Jakštys
• [asterisk-users] Jack in /usr/local/ means failure for asterisk   Motiejus Jakštys
• [asterisk-users] Help with IP Routing   Motiejus Jakštys
• [asterisk-users] VoIP over virtualized VPN   Motiejus Jakštys
• [asterisk-users] Jack in /usr/local/ means failure for asterisk   Motiejus Jakštys
• [asterisk-users] VoIP over virtualized VPN   Motiejus Jakštys
• [asterisk-users] Jack in /usr/local/ means failure for asterisk   Motiejus Jakštys
• [asterisk-users] Jack in /usr/local/ means failure for asterisk   Motiejus Jakštys
• [asterisk-users] Dialing a SIP Peer without using register strin   Nasir Javaid
• [asterisk-users] asterisk-users Digest, Vol 70, Issue 23   Nasir Javaid
• [asterisk-users] asterisk-users Digest, Vol 70, Issue 24   Nasir Javaid
• [asterisk-users] asterisk-users Digest, Vol 70, Issue 25   Nasir Javaid
• [asterisk-users] asterisk-users Digest, Vol 70, Issue 25   Nasir Javaid
• [asterisk-users] asterisk-users Digest, Vol 70, Issue 25   Nasir Javaid
• [asterisk-users] asterisk-users Digest, Vol 70, Issue 30   Nasir Javaid
• [asterisk-users] asterisk-users Digest, Vol 70, Issue 54   Nasir Javaid
• [asterisk-users] Switchvox vs Asterisk codebase   Andrew Joakimsen
• [asterisk-users] Asterisk Call Recording *1 Status Indication   Steve Johnson
• [asterisk-users] [Asterisk-Users] Asterisk transfer to a conference using feature code?   Steve Johnson
• [asterisk-users] How to get ConfBridge user count   Steve Johnson
• [asterisk-users] GoogleTalk to Asterisk - choosing voice menu options   Steve Johnson
• [asterisk-users] iax calls via checkbox.cc   Joseph
• [asterisk-users] iax calls via checkbox.cc   Joseph
• [asterisk-users] iax calls via checkbox.cc   Joseph
• [asterisk-users] iax calls via checkbox.cc   Joseph
• [asterisk-users] IAX2 - providers discontinuing support   Joseph
• [asterisk-users] IAX2 - providers discontinuing support   Joseph
• [asterisk-users] Calls Dropping   Dan Journo
• [asterisk-users] Calls Dropping   Dan Journo
• [asterisk-users] Reading the CDR   Dan Journo
• [asterisk-users] Reading the CDR   Dan Journo
• [asterisk-users] Reading the CDR   Dan Journo
• [asterisk-users] text   Dan Journo
• [asterisk-users] Ringback   Dan Journo
• [asterisk-users] Problems with Asterisk and two Linksys SPA941   Dan Journo
• [asterisk-users] Portech MV-374 does not register behind NAT   Jonas Kellens
• [asterisk-users] DSCP QoS value in YeaLink phone settings   Jonas Kellens
• [asterisk-users] No ringtone when going from queue to dial-command   Jonas Kellens
• [asterisk-users] No ringtone when going from queue to dial-command   Jonas Kellens
• [asterisk-users] No ringtone when going from queue to dial-command   Jonas Kellens
• [asterisk-users] app_addon_sql_mysql.c:116 find_identifier   Jonas Kellens
• [asterisk-users] app_addon_sql_mysql.c:116 find_identifier   Jonas Kellens
• [asterisk-users] No ringtone when going from queue to dial-command   Jonas Kellens
• [asterisk-users] app_addon_sql_mysql.c:116 find_identifier   Jonas Kellens
• [asterisk-users] Asterisk 1.4.30 & T38   Jonas Kellens
• [asterisk-users] Sending fake auth rejection for user   Jonas Kellens
• [asterisk-users] Hanging up call - no reply to our critical packet   Jonas Kellens
• [asterisk-users] Hanging up call - no reply to our critical packet   Jonas Kellens
• [asterisk-users] shared lines (sla) with Asterisk 1.4.26, any hints?   Jonas Kellens
• [asterisk-users] Voicemail : mail attachment to multiple mail-addresses   Jonas Kellens
• [asterisk-users] NPA NXX Database   Don Kelly
• [asterisk-users] Asterisk 1.6.2.7 Now Available   Richard Kenner
• [asterisk-users] Asterisk 1.6.2.7 Now Available   Richard Kenner
• [asterisk-users] Possible bug in chan_sip:add_sdp   Richard Kenner
• [asterisk-users] Possible bug in chan_sip:add_sdp   Richard Kenner
• [asterisk-users] Video in Skype for Asterisk   Richard Kenner
• [asterisk-users] Getting presence working in 1.6.2   Richard Kenner
• [asterisk-users] Working with Blacklist database   Richard Kenner
• [asterisk-users] Speech/DTMF mix?   Richard Kenner
• [asterisk-users] Speech/DTMF mix?   Richard Kenner
• [asterisk-users] Speech/DTMF mix?   Richard Kenner
• [asterisk-users] Speech/DTMF mix?   Richard Kenner
• [asterisk-users] Speech/DTMF mix?   Richard Kenner
• [asterisk-users] Digits and Vestec   Richard Kenner
• [asterisk-users] Digits and Vestec   Richard Kenner
• [asterisk-users] Possible bug in chan_sip:add_sdp   Richard Kenner
• [asterisk-users] Delay on DTMF with SpeechBackground and Vestec   Richard Kenner
• [asterisk-users] voice recognition suggestion   Richard Kenner
• [asterisk-users] Normalizing called numbers   Richard Kenner
• [asterisk-users] Wierd behavior of illegal extension   Richard Kenner
• [asterisk-users] Wierd behavior of illegal extension   Richard Kenner
• [asterisk-users] bug in asterisk   Mindaugas Kezys
• [asterisk-users] bug in asterisk   Mindaugas Kezys
• [asterisk-users] Hangup Detection   Shariq Khan
• [asterisk-users] AMR codec for Asterisk 1.6.1.X   Kyle Kienapfel
• [asterisk-users] RTP ports   Philipp von Klitzing
• [asterisk-users] Reading the CDR   Philipp von Klitzing
• [asterisk-users] What is billsec in CDR?   Philipp von Klitzing
• [asterisk-users] Hash Dial Pattern Problems   Philipp von Klitzing
• [asterisk-users] Make the call finish after executing Dial(G())   Philipp von Klitzing
• [asterisk-users] OT: NAT in SPA922   Philipp von Klitzing
• [asterisk-users] Speech/DTMF mix?   Philipp von Klitzing
• [asterisk-users] SIP trunk between two Asterisk servers   Philipp von Klitzing
• [asterisk-users] SIP trunk between two Asterisk servers   Philipp von Klitzing
• [asterisk-users] SIP trunk between two Asterisk servers   Philipp von Klitzing
• [asterisk-users] SIP trunk between two Asterisk servers   Philipp von Klitzing
• [asterisk-users] SIP trunk between two Asterisk servers [SOLVED]   Philipp von Klitzing
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Philipp von Klitzing
• [asterisk-users] About option U in Dial Ast version 1.6.2   Philipp von Klitzing
• [asterisk-users] mISDN compiling error   Philipp von Klitzing
• [asterisk-users] OpenVox B200P and D410P under Asterisk 1.6   Philipp von Klitzing
• [asterisk-users] sending T.38 fax negotiation problem   Ilmars Knipshis
• [asterisk-users] working example of t38 fax w/ 1.6.2?   Ilmars Knipšis
• [asterisk-users] play a sound file directly to a caller channel   Daniel Knoll
• [asterisk-users] play a sound file directly to a caller channel   Daniel Knoll
• [asterisk-users] call droped if second caller enter meetme conference   Daniel Knoll
• [asterisk-users] meetme changes between asterisk 1.6.2.6 and 1.6.2.7   Daniel Knoll
• [asterisk-users] TTS for asterisk   Stelios Koroneos
• [asterisk-users] zap calls are getting dropped (unexpected disconnect message)   Rustam Kovhaev
• [asterisk-users] [0017330] 1.6.1 and 1.6.2 + MySQL crases on ODBC Query (via func_odbc or sip realtime)   Marcin J. Kowalczyk
• [asterisk-users] Read and set the UUI in asterisk   velusamy Krishnan
• [asterisk-users] HDLC Receiver overrun on Wildcard TE410P   Łukasz Krzyżak
• [asterisk-users] Help with IP Routing   Nivin Kumar
• [asterisk-users] Help with IP Routing   Nivin Kumar
• [asterisk-users] OT: Help with IP Routing   Nivin Kumar
• [asterisk-users] Connect mobile to asterisk   Nivin Kumar
• [asterisk-users] Switchvox vs Asterisk codebase   Nivin Kumar
• [asterisk-users] VoIP Termination in Japan   Andy Kuo
• [asterisk-users] VoIP Termination in Japan   Andy Kuo
• [asterisk-users] Suddenly "HDLC Bad FCS (8)" errors on ISDN-PRI, changed nothing   DLeese at LStelcom.com
• [asterisk-users] OT: NAT in SPA922   Jeff LaCoursiere
• [asterisk-users] OT: NAT in SPA922   Jeff LaCoursiere
• [asterisk-users] Digits and Vestec   Jeff LaCoursiere
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Jeff LaCoursiere
• [asterisk-users] "ring splash"   Jeff LaCoursiere
• [asterisk-users] VoIP over virtualized VPN   Jeff LaCoursiere
• [asterisk-users] "ring splash"   Jeff LaCoursiere
• [asterisk-users] How to have Asterisk respond from the IP address used for registration   Jeff LaCoursiere
• [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286   James Lamanna
• [asterisk-users] Registering a Cisco 7965 on 1.4.26   James Lamanna
• [asterisk-users] Registering a Cisco 7965 on 1.4.26   James Lamanna
• [asterisk-users] OT: NAT in SPA922   James Lamanna
• [asterisk-users] OT: NAT in SPA922   James Lamanna
• [asterisk-users] Extension state can get stuck in 'Ringing' state   James Lamanna
• [asterisk-users] Asterisk 1.4.30 & T38   Travis Langhals
• [asterisk-users] Asterisk 1.4.30 & T38   Travis Langhals
• [asterisk-users] Asterisk Bible?   Andrew Latham
• [asterisk-users] Using Asterisk? Get on the press list!   Andrew Latham
• [asterisk-users] Softphones on thin clients...   Andrew Latham
• [asterisk-users] "Contact header appears incorrect on this invite" Asterisk registering with another PBX   Mike A. Leonetti
• [asterisk-users] "Contact header appears incorrect on this invite" Asterisk registering with another PBX   Mike A. Leonetti
• [asterisk-users] "Contact header appears incorrect on this invite" Asterisk registering with another PBX   Mike A. Leonetti
• [asterisk-users] "Contact header appears incorrect on this invite" Asterisk registering with another PBX   Mike A. Leonetti
• [asterisk-users] "Contact header appears incorrect on this invite" Asterisk registering with another PBX   Mike A. Leonetti
• [asterisk-users] SIP REGISTER header not containing Allow-Events or Allow   Mike A. Leonetti
• [asterisk-users] Sending SIP credentials in INVITE   Mike A. Leonetti
• [asterisk-users] FollowMe dials numbers but can't confirm the call or hear anything   Mike A. Leonetti
• [asterisk-users] CDR to MS-SQL via ODBC issue   Tilghman Lesher
• [asterisk-users] CDR to MS-SQL via ODBC issue   Tilghman Lesher
• [asterisk-users] Getting presence working in 1.6.2   Tilghman Lesher
• [asterisk-users] Getting presence working in 1.6.2   Tilghman Lesher
• [asterisk-users] Getting presence working in 1.6.2   Tilghman Lesher
• [asterisk-users] Records sets and ODBC   Tilghman Lesher
• [asterisk-users] 1.6.2.7 SIP realtime problem   Tilghman Lesher
• [asterisk-users] ISDN config: LBO values   Tilghman Lesher
• [asterisk-users] ISDN config: LBO values   Tilghman Lesher
• [asterisk-users] Adding a context from the console   Tilghman Lesher
• [asterisk-users] Little t38 bug?   Tilghman Lesher
• [asterisk-users] Adding a context from the console   Tilghman Lesher
• [asterisk-users] Adding a context from the console   Tilghman Lesher
• [asterisk-users] DAHDI Help (made a cardinal sin :()   Tilghman Lesher
• [asterisk-users] Wierd behavior of illegal extension   Tilghman Lesher
• [asterisk-users] Wierd behavior of illegal extension   Tilghman Lesher
• [asterisk-users] AGI <==> DeadAGI   Luki
• [asterisk-users] OT: NAT in SPA922   Luki
• [asterisk-users] Channels In Use   Luki
• [asterisk-users] Forwarding inbound mobiles   Julian Lyndon-Smith
• [asterisk-users] Confirm answering a call   Doug Lytle
• [asterisk-users] Need fax solution for 1.4.xx   Doug Lytle
• [asterisk-users] Lookup ${EXTEN} in database, update context/route if found... AGI?   Doug Lytle
• [asterisk-users] app_addon_sql_mysql.c:116 find_identifier   Doug Lytle
• [asterisk-users] app_addon_sql_mysql.c:116 find_identifier   Doug Lytle
• [asterisk-users] app_addon_sql_mysql.c:116 find_identifier   Doug Lytle
• [asterisk-users] convert zaptel to dahdi?   Doug Lytle
• [asterisk-users] routing of calls   Doug Lytle
• [asterisk-users] routing of calls   Doug Lytle
• [asterisk-users] sending T.38 fax negotiation problem   Leif Madsen
• [asterisk-users] Asterisk 1.6.2.7 Now Available   Leif Madsen
• [asterisk-users] Asterisk 1.6.2.7 Now Available   Leif Madsen
• [asterisk-users] Asterisk 1.6.2.7 Now Available   Leif Madsen
• [asterisk-users] CDR to MS-SQL via ODBC issue   Leif Madsen
• [asterisk-users] Lookup ${EXTEN} in database, update context/route if found... AGI?   Leif Madsen
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Leif Madsen
• [asterisk-users] Which issue is keeping you from updrading to1.6.2 ?   Leif Madsen
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Leif Madsen
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Leif Madsen
• [asterisk-users] Cause and cure for "Exceptionally long voice queue length queuing to Local"?   Leif Madsen
• [asterisk-users] Asterisk 1.4.31 IAX2/RSA BUG   Andre Magalhaes
• [asterisk-users] Reading the CDR   Ishfaq Malik
• [asterisk-users] No ringtone when going from queue to dial-command   Ishfaq Malik
• [asterisk-users] VoIP Termination in Japan   Adrian Marsh
• [asterisk-users] AMR codec for Asterisk 1.6.1.X   Adrian Marsh
• [asterisk-users] Questions About Fax for Asterisk   Martin
• [asterisk-users] DAHDI and ESXi   Conor McTernan
• [asterisk-users] DAHDI and ESXi   Conor McTernan
• [asterisk-users] Lookup ${EXTEN} in database, update context/route if found... AGI?   Anthony Messina
• [asterisk-users] BRI card(B800P) doesn's work with DAHDI(wcb4xxp) in NT mode   Michael
• [asterisk-users] BRI card(B800P) doesn's work with DAHDI(wcb4xxp) in NT mode   Michael
• [asterisk-users] new way of asterisk and kamailio (openser) realtime integration   Daniel-Constantin Mierla
• [asterisk-users] R: new way of asterisk and kamailio (openser) realtime integration   Daniel-Constantin Mierla
• [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration   Daniel-Constantin Mierla
• [asterisk-users] Question about 1.6: multiple IP on a single Asterisk box / multi ISP routing   Mike
• [asterisk-users] OT: Windows TAPI command-line driver   Mike
• [asterisk-users] OT: Windows TAPI command-line driver   Mike
• [asterisk-users] How to have Asterisk respond from the IP address used for registration   Mike
• [asterisk-users] How to have Asterisk respond from the IP addressused for registration   Mike
• [asterisk-users] How to have Asterisk respond from the IP address used for registration   Mike
• [asterisk-users] How to have Asterisk respond from the IP address used for registration   Mike
• [asterisk-users] How to use one single IP as origination   Mike
• [asterisk-users] How to use one single IP as origination   Mike
• [asterisk-users] How to use one single IP as origination   Mike
• [asterisk-users] Reloading queue members (realtime DB)   Mike
• [asterisk-users] Reloading queue members (realtime DB)   Mike
• [asterisk-users] Voicemail : mail attachment to multiple mail-addresses   Mike
• [asterisk-users] How to have Asterisk respond from the IP address used for registration   Mike
• [asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server   Eddie Mikell
• [asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server - it is robbery!   Eddie Mikell
• [asterisk-users] Multiple SIP lines.   Eddie Mikell
• [asterisk-users] More clarification on outbound sip channels.   Eddie Mikell
• [asterisk-users] Stress Test new system   Eddie Mikell
• [asterisk-users] Pattern matching - how to ignore numbers after 10 digits   Eddie Mikell
• [asterisk-users] OT: NAT in SPA922   Sebastian Milioto
• [asterisk-users] OT: NAT in SPA922   Sebastian Milioto
• [asterisk-users] OT: NAT in SPA922   Sebastian Milioto
• [asterisk-users] OT: NAT in SPA922   Sebastian Milioto
• [asterisk-users] OT: NAT in SPA922   Sebastian Milioto
• [asterisk-users] iax calls via checkbox.cc   Barry Miller
• [asterisk-users] Transfer calls using ##   Noah Miller
• [asterisk-users] Asterisk Query   Noah Miller
• [asterisk-users] OT: NAT in SPA922   Noah Miller
• [asterisk-users] OT: NAT in SPA922   Noah Miller
• [asterisk-users] Help with IP Routing   Adam Moffett
• [asterisk-users] Need fax solution for 1.4.xx   Miguel Molina
• [asterisk-users] run script after completed   Mickael Monsieur
• [asterisk-users] Cant load chan_dahdi   Enrique Mora
• [asterisk-users] Cant load chan_dahdi   Enrique Mora
• [asterisk-users] Cant load chan_dahdi   Enrique Mora
• [asterisk-users] Skype for Asterisk and instant messages   Enrique Mora
• [asterisk-users] problem with trustrpid   Dan Moschuk
• [asterisk-users] convert zaptel to dahdi?   Michael Munger
• [asterisk-users] file command with alaw file   Steve Murphy
• [asterisk-users] file command with alaw file   Steve Murphy
• [asterisk-users] Attended Transfer using AMI   Grant Murray
• [asterisk-users] Interesting email project.   Tim Nelson
• [asterisk-users] Channel failover   Tim Nelson
• [asterisk-users] Lookup ${EXTEN} in database, update context/route if found... AGI?   Tim Nelson
• [asterisk-users] Sangoma A101D PRI failing with ERROR - -- Got SABME from network peer. Sending Unnumbered Acknowledgement   Tim Nelson
• [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality   Tim Nelson
• [asterisk-users] About Sangoma cards and Asterisk integration with other PBX   Tim Nelson
• [asterisk-users] Help with IP Routing   Tim Nelson
• [asterisk-users] "ring splash"   Tim Nelson
• [asterisk-users] "pri show version" still shows old version despite doing a make && make clean && make install for v1.4.11   Tim Nelson
• [asterisk-users] DAHDI Help (made a cardinal sin :()   Tim Nelson
• [asterisk-users] Calls Dropping   matthieu Nicaise
• [asterisk-users] Parking problem with outgoing calls   matthieu Nicaise
• [asterisk-users] RTP ports   Danny Nicholas
• [asterisk-users] BADTIME FOR ANSWEREDTIME   Danny Nicholas
• [asterisk-users] Bridging old system (ESI IVX E) with new Asteriskserver   Danny Nicholas
• [asterisk-users] Run a script after Page application   Danny Nicholas
• [asterisk-users] Code in extensions.conf to leave a voice mailin another PBX ?!   Danny Nicholas
• [asterisk-users] Hash Dial Pattern Problems   Danny Nicholas
• [asterisk-users] Code in extensions.conf to leave a voicemailin another PBX ?!   Danny Nicholas
• [asterisk-users] Hash Dial Pattern Problems   Danny Nicholas
• [asterisk-users] Hash Dial Pattern Problems   Danny Nicholas
• [asterisk-users] Hash Dial Pattern Problems   Danny Nicholas
• [asterisk-users] Registering a Cisco 7965 on 1.4.26   Danny Nicholas
• [asterisk-users] run script after completed   Danny Nicholas
• [asterisk-users] Hash Dial Pattern Problems   Danny Nicholas
• [asterisk-users] Hash Dial Pattern Problems   Danny Nicholas
• [asterisk-users] Getting presence working in 1.6.2   Danny Nicholas
• [asterisk-users] Getting presence working in 1.6.2   Danny Nicholas
• [asterisk-users] Execute AGI, Then Continue   Danny Nicholas
• [asterisk-users] DAHDI not detecting hangup   Danny Nicholas
• [asterisk-users] DAHDI not detecting hangup   Danny Nicholas
• [asterisk-users] Speech/DTMF mix?   Danny Nicholas
• [asterisk-users] AGI and Severe Weather Alerts   Danny Nicholas
• [asterisk-users] Speech/DTMF mix?   Danny Nicholas
• [asterisk-users] Speech/DTMF mix?   Danny Nicholas
• [asterisk-users] AGI and Severe Weather Alerts   Danny Nicholas
• [asterisk-users] Need fax solution for 1.4.xx   Danny Nicholas
• [asterisk-users] Digits and Vestec   Danny Nicholas
• [asterisk-users] Help finding online training   Danny Nicholas
• [asterisk-users] Stress Test new system   Danny Nicholas
• [asterisk-users] Play MusicOnHold and continue with dialplan   Danny Nicholas
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Danny Nicholas
• [asterisk-users] Play MusicOnHold and continue with dialplan   Danny Nicholas
• [asterisk-users] DTMF Input from the User   Danny Nicholas
• [asterisk-users] Cause and cure for "Exceptionally long voice queuelength queuing to Local"?   Danny Nicholas
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Danny Nicholas
• [asterisk-users] file command with alaw file   Danny Nicholas
• [asterisk-users] Checking blank CallerID in Dialplan   Danny Nicholas
• [asterisk-users] Which issue is keeping you from updrading to1.6.2 ?   Danny Nicholas
• [asterisk-users] file command with alaw file   Danny Nicholas
• [asterisk-users] SPA-932 BLF Stays solid red for single extension   Danny Nicholas
• [asterisk-users] SPA-932 BLF Stays solid red for singleextension   Danny Nicholas
• [asterisk-users] TTS for asterisk   Danny Nicholas
• [asterisk-users] sip and SSL   Danny Nicholas
• [asterisk-users] convert zaptel to dahdi?   Danny Nicholas
• [asterisk-users] routing of calls   Danny Nicholas
• [asterisk-users] OT: Windows TAPI command-line driver   Danny Nicholas
• [asterisk-users] Getting 'username' of sip peer   Danny Nicholas
• [asterisk-users] Music on Hold   Danny Nicholas
• [asterisk-users] Music on Hold   Danny Nicholas
• [asterisk-users] Adding a context from the console   Danny Nicholas
• [asterisk-users] How to have Asterisk respond from the IP addressused for registration   Danny Nicholas
• [asterisk-users] Normalizing called numbers   Danny Nicholas
• [asterisk-users] Normalizing called numbers   Danny Nicholas
• [asterisk-users] Hash Dial Pattern Problems   David Nickel
• [asterisk-users] Hash Dial Pattern Problems   David Nickel
• [asterisk-users] Hash Dial Pattern Problems   David Nickel
• [asterisk-users] Hash Dial Pattern Problems   David Nickel
• [asterisk-users] Hash Dial Pattern Problems   David Nickel
• [asterisk-users] Hash Dial Pattern Problems   David Nickel
• [asterisk-users] Hash Dial Pattern Problems   David Nickel
• [asterisk-users] Hash Dial Pattern Problems   David Nickel
• [asterisk-users] nortel meridian question   Dale Noll
• [asterisk-users] AGI <==> DeadAGI   John Novack
• [asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server   John Novack
• [asterisk-users] DAHDI not detecting hangup   John Novack
• [asterisk-users] iax calls via checkbox.cc   John Novack
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   John Novack
• [asterisk-users] Checking blank CallerID in Dialplan   John Novack
• [asterisk-users] About Sangoma cards and Asterisk integration with other PBX   John Novack
• [asterisk-users] Queue ringall problem.   Massimo Nuvoli
• [asterisk-users] working example of t38 fax w/ 1.6.2?   Olivier
• [asterisk-users] JNET's qozap, dahdi and PCI-E Quad   Olivier
• [asterisk-users] JNET's qozap, dahdi and PCI-E Quad   Olivier
• [asterisk-users] OT - Newbie: Do I have a shared IRQ ?   Olivier
• [asterisk-users] Conflict between jumper settings and dmesg with OctoBRI   Olivier
• [asterisk-users] Bug or feature: comments in chan_dahdi.conf.sample   Olivier
• [asterisk-users] B410P and Patton smartnode : any success ?   Olivier
• [asterisk-users] JNET's qozap, dahdi and PCI-E Quad   Olivier
• [asterisk-users] identify caller hangup or callee hangup?   Olivier
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Olivier
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Olivier
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Olivier
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   Olivier
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Olivier
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Olivier
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Olivier
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Olivier
• [asterisk-users] OT - How to query vcard-like data for CTI app   Olivier
• [asterisk-users] Is AstManProxy still recommended with 1.6 and later ?   Olivier
• [asterisk-users] Can't specify connection port with ADA   Olivier
• [asterisk-users] ADA 1.1 features   Olivier
• [asterisk-users] R: new way of asterisk and kamailio (openser) realtime integration   Alexandru Oniciuc
• [asterisk-users] Little t38 bug?   Alexandru Oniciuc
• [asterisk-users] B400P card crashes conncection   Rudi Oosthuizen
• [asterisk-users] Issues with remote call setup   Vinod Parameswaran
• [asterisk-users] Re-invite from Asterisk Server: Port number changes   Vinod Parameswaran
• [asterisk-users] include sip configuration from another file in sip.conf   Jason Parker
• [asterisk-users] Dahdi problems with kernel 2.6.32   Jason Parker
• [asterisk-users] text   Thomas Perron
• [asterisk-users] text   Thomas Perron
• [asterisk-users] Simulating a commercial SIP provider   Alfredo Peña
• [asterisk-users] Normalizing called numbers   Rémi Philippe
• [asterisk-users] Normalizing called numbers   Rémi Philippe
• [asterisk-users] IAX2 Call Transfer   Wolfgang Pichler
• [asterisk-users] B410P and Patton smartnode : any success ?   Leonardo Pistone
• [asterisk-users] Sip session timers.   Leonardo Pistone
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Leonardo Pistone
• [asterisk-users] R: new way of asterisk and kamailio (openser) realtime integration   Fred Posner
• [asterisk-users] NPA NXX Database   Fred Posner
• [asterisk-users] Checking blank CallerID in Dialplan   Fred Posner
• [asterisk-users] Peering with a Taqua T7000   Philip Prindeville
• [asterisk-users] US "Truth in caller id act"... and it's impact on services   Philip Prindeville
• [asterisk-users] US "Truth in caller id act"... and it's impact onservices   Philip Prindeville
• [asterisk-users] Asterisk 1.6.2.7 + app_fax + OpenBSD 4.7 minor issue   Noah Pugsley
• [asterisk-users] Recording with extension and agent in queue   Edwin Quijada
• [asterisk-users] voipmonitor.org   Edwin Quijada
• [asterisk-users] Error at start of asterisk with cdr_addon_mysql.o   Pham Quy
• [asterisk-users] Error at start of asterisk with cdr_addon_mysql.o   Pham Quy
• [asterisk-users] file command with alaw file   Pham Quy
• [asterisk-users] file command with alaw file   Pham Quy
• [asterisk-users] file command with alaw file   Pham Quy
• [asterisk-users] file command with alaw file   Pham Quy
• [asterisk-users] file command with alaw file   Pham Quy
• [asterisk-users] Being attacked by an Amazon EC2 ...   Randy R
• [asterisk-users] RTP ports   Randy R
• [asterisk-users] RTP ports   Randy R
• [asterisk-users] R: new way of asterisk and kamailio (openser) realtime integration   Randy R
• [asterisk-users] Friday @12 Noon and 1PM   Randy R
• [asterisk-users] Asterisk-based "Incredible PBX"   Randy R
• [asterisk-users] SIP - SIP over PBX no audio when canreinvite=no   RG
• [asterisk-users] Hi   Rajkiran Reddy
• [asterisk-users] app_page.so was missing   John Regal
• [asterisk-users] IAX2 Auto-congesting call due to slow response   Alexandre Rodrigues
• [asterisk-users] IAX2 Auto-congesting call due to slow response   Alexandre Rodrigues
• [asterisk-users] "bad magic number" log messages   John Rose
• [asterisk-users] "bad magic number" log messages   John Rose
• [asterisk-users] "bad magic number" log messages   John Rose
• [asterisk-users] [0017330] 1.6.1 and 1.6.2 + MySQL crases onODBC Query (via func_odbc or sip realtime)   John Rose
• [asterisk-users] Better AMD module   John Rose
• [asterisk-users] Better AMD module   John Rose
• [asterisk-users] AGI <==> DeadAGI   SIP
• [asterisk-users] AGI <==> DeadAGI   SIP
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   SIP
• [asterisk-users] OT - Which SIP hardphone with 100 BLF ?   SIP
• [asterisk-users] Sometimes called party answers, but callee keep hear ringing, called party hears nothing!   Raimund Sacherer
• [asterisk-users] Xorcom Astribank One call and dies   Raimund Sacherer
• [asterisk-users] E1 information   Felipe Conde Sales
• [asterisk-users] Re TrixBox   Samantha
• [asterisk-users] Delay in IVR   Sasa
• [asterisk-users] a2billing DID and Queues   Tarek Sawah
• [asterisk-users] Asterisk and RFC 3261   Tarek Sawah
• [asterisk-users] Queue ringall problem.   Tarek Sawah
• [asterisk-users] Queue ringall problem.   Tarek Sawah
• [asterisk-users] CANCEL Reason   Stefan Schmidt
• [asterisk-users] Music on Hold   Stefan Schmidt
• [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286   Mark Scholten
• [asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server   Mark Scholten
• [asterisk-users] Confirm answering a call   Mark Scholten
• [asterisk-users] Help with IP Routing   Roger Schreiter
• [asterisk-users] Help with IP Routing   Roger Schreiter
• [asterisk-users] Help finding online training   Joseph Schwartz
• [asterisk-users] IAX2 Load test   Jon Schøpzinsky
• [asterisk-users] Continue dialplan is source channel hangs up   Sebastian
• [asterisk-users] Registering a Cisco 7965 on 1.4.26   Warren Selby
• [asterisk-users] T.38 Fax With Flowroute SIP Provider   Warren Selby
• [asterisk-users] SPA-932 BLF Stays solid red for single extension   Warren Selby
• [asterisk-users] Error compiling DAHDI...   Warren Selby
• [asterisk-users] Connecting 1-2 GSM ports to asterisk?   Hugo Serrano
• [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension   Zhang Shukun
• [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension   Zhang Shukun
• [asterisk-users] Creating a HTTP Request on missed call?   Zhang Shukun
• [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension   Zhang Shukun
• [asterisk-users] Creating a HTTP Request on missed call?   Zhang Shukun
• [asterisk-users] Could Asterisk PHP agi be a SOAP Client?   Zhang Shukun
• [asterisk-users] is my PHPAGI Soap code right?   Zhang Shukun
• [asterisk-users] identify caller hangup or callee hangup?   Zhang Shukun
• [asterisk-users] is my PHPAGI Soap code right?   Zhang Shukun
• [asterisk-users] Why Manager account log on and log off alternatively all the time?   Zhang Shukun
• [asterisk-users] SuSE Firewall2 - Port Forward Command   Marco Signorini
• [asterisk-users] automon filename does not follow the docs.   Marta Silva
• [asterisk-users] automon filename does not follow the docs.   Marta Silva
• [asterisk-users] problems with unicall   Moises Silva
• [asterisk-users] Getting "ghost" transfer or music on hold   Prince Singh
• [asterisk-users] identify caller hangup or callee hangup?   Aurimas Skirgaila
• [asterisk-users] Getting presence working in 1.6.2   Jared Smith
• [asterisk-users] How to get ConfBridge user count   Jared Smith
• [asterisk-users] Getting 'username' of sip peer   Jared Smith
• [asterisk-users] Calls Dropping   Bob Smither
• [asterisk-users] Calls Dropping   Bob Smither
• [asterisk-users] OpenVox B200P and D410P under Asterisk 1.6   Scott Stingel
• [asterisk-users] OpenVox B200P and D410P under Asterisk 1.6   Scott Stingel
• [asterisk-users] DID's for Chatham, ON   Scott Stingel
• [asterisk-users] Run a script after Page application   Andy Swing
• [asterisk-users] Spy on Asterisk 1.2   Torintino T
• [asterisk-users] Connecting 1-2 GSM ports to asterisk?   Administrator TOOTAI
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Kingsley Tart
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Kingsley Tart
• [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script   Kingsley Tart
• [asterisk-users] Delay in IVR   Kingsley Tart
• [asterisk-users] Asterisk 1.4.31 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.0.27 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.1.19 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.2.7 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.0.28 and 1.6.1.20 Now Available   Asterisk Development Team
• [asterisk-users] DAHDI Linux 2.3.0.1 and DAHDI Linux 2.2.1.2 Released   Asterisk Development Team
• [asterisk-users] Libpri 1.4.11 Released   Asterisk Development Team
• [asterisk-users] Asterisk Cluster   Jonathan Thurman
• [asterisk-users] Asterisk Cluster   Jonathan Thurman
• [asterisk-users] Best way to limit outgoing calls per trunk   Jonathan Thurman
• [asterisk-users] Best way to limit outgoing calls per trunk   Jonathan Thurman
• [asterisk-users] Using Asterisk? Get on the press list!   John Todd
• [asterisk-users] run extensions after call moved to queue and answered by member   Vasiliy G Tolstov
• [asterisk-users] run extensions after call moved to queue and answered by member   Vasiliy G Tolstov
• [asterisk-users] SuSE Firewall2 - Port Forward Command   Brent A. Torrenga
• [asterisk-users] Questions About Fax for Asterisk   Steve Totaro
• [asterisk-users] Questions About Fax for Asterisk   Steve Totaro
• [asterisk-users] Questions About Fax for Asterisk   Steve Totaro
• [asterisk-users] Questions About Fax for Asterisk   Steve Totaro
• [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality   Steve Totaro
• [asterisk-users] Asterisk Crashing with ERROR[1906] astobj2.c: refcount -1 on object 0xb1aab758 Ast Ver 1.6.2.6   Steve Totaro
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Steve Totaro
• [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286   Steve Underwood
• [asterisk-users] Questions About Fax for Asterisk   Steve Underwood
• [asterisk-users] Need fax solution for 1.4.xx   Steve Underwood
• [asterisk-users] Need fax solution for 1.4.xx (Resolution)   Steve Underwood
• [asterisk-users] Little t38 bug?   Steve Underwood
• [asterisk-users] dialplan question   Vardan
• [asterisk-users] Dialing a SIP Peer without using register strin   Vardan
• [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension   Vardan
• [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension   Vardan
• [asterisk-users] asterisk-users Digest, Vol 70, Issue 23   Vardan
• [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension   Vardan
• [asterisk-users] bug in asterisk   Vardan
• [asterisk-users] asterisk-users Digest, Vol 70, Issue 24   Vardan
• [asterisk-users] No ringtone when going from queue to dial-command   Vardan
• [asterisk-users] No ringtone when going from queue to dial-command   Vardan
• [asterisk-users] SIP trunk between two Asterisk servers   Vardan
• [asterisk-users] SIP trunk between two Asterisk servers   Vardan
• [asterisk-users] SIP trunk between two Asterisk servers   Vardan
• [asterisk-users] SIP trunk between two Asterisk servers   Vardan
• [asterisk-users] SIP trunk between two Asterisk servers   Vardan
• [asterisk-users] SIP trunk between two Asterisk servers   Vardan
• [asterisk-users] SIP trunk between two Asterisk servers   Vardan
• [asterisk-users] asterisk-users Digest, Vol 70, Issue 25   Vardan
• [asterisk-users] SIP trunk between two Asterisk servers   Vardan
• [asterisk-users] 1.6.2.7 SIP realtime problem   Vardan
• [asterisk-users] identify caller hangup or callee hangup?   Vardan
• [asterisk-users] identify caller hangup or callee hangup?   Vardan
• [asterisk-users] identify caller hangup or callee hangup?   Vardan
• [asterisk-users] R: new way of asterisk and kamailio (openser) realtime integration   Vardan
• [asterisk-users] identify caller hangup or callee hangup?   Vardan
• [asterisk-users] About option U in Dial Ast version 1.6.2   Vardan
• [asterisk-users] a2billing DID and Queues   Vardan
• [asterisk-users] About option U in Dial Ast version 1.6.2   Vardan
• [asterisk-users] queue members   Vieri
• [asterisk-users] asterisk and gnokii on same server: scratchy sound   Vieri
• [asterisk-users] queue member state in asterisk 1.4   Vieri
• [asterisk-users] SIP trunk between two Asterisk servers   Vieri
• [asterisk-users] SIP trunk between two Asterisk servers   Vieri
• [asterisk-users] SIP trunk between two Asterisk servers   Vieri
• [asterisk-users] SIP trunk between two Asterisk servers   Vieri
• [asterisk-users] SIP trunk between two Asterisk servers   Vieri
• [asterisk-users] SIP trunk between two Asterisk servers   Vieri
• [asterisk-users] SIP trunk between two Asterisk servers   Vieri
• [asterisk-users] SIP trunk between two Asterisk servers   Vieri
• [asterisk-users] SIP trunk between two Asterisk servers   Vieri
• [asterisk-users] SIP trunk between two Asterisk servers [SOLVED]   Vieri
• [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality   Vieri
• [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality   Vieri
• [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality   Vieri
• [asterisk-users] SIP trunk between two Asterisk servers [SOLVED]   Vieri
• [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality   Vieri
• [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality   Vieri
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Vincent
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Vincent
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Vincent
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Vincent
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Vincent
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Vincent
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Vincent
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Vincent
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Vincent
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Vincent
• [asterisk-users] [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?   Vincent
• [asterisk-users] [X100P+Dahdi 2.3.0] Couple of questions   Vincent
• [asterisk-users] [Dahdi/system.conf] "fxsks = 1" deprecated?   Vincent
• [asterisk-users] [Dahdi/system.conf] "fxsks = 1" deprecated?   Vincent
• [asterisk-users] [Dahdi/system.conf] "fxsks = 1" deprecated?   Vincent
• [asterisk-users] [Dahdi/system.conf] "fxsks = 1" deprecated?   Vincent
• [asterisk-users] Connect mobile to asterisk   Vincent
• [asterisk-users] Connect mobile to asterisk   Vincent
• [asterisk-users] voipmonitor.org   Martin Vit
• [asterisk-users] voipmonitor.org   Martin Vit
• [asterisk-users] voipmonitor.org   Martin Vit
• [asterisk-users] Call Timeout on 302 Redirect   Jay Vocaire
• [asterisk-users] voipmonitor.org   Martin Vít
• [asterisk-users] include sip configuration from another file in sip.conf   Robert Wagner
• [asterisk-users] pattern containing an asterisk   Robert Wagner
• [asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server   Ryan Wagoner
• [asterisk-users] T.38 Fax With Flowroute SIP Provider   Ryan Wagoner
• [asterisk-users] T.38 Fax With Flowroute SIP Provider   Ryan Wagoner
• [asterisk-users] T.38 Fax With Flowroute SIP Provider   Ryan Wagoner
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Ryan Wagoner
• [asterisk-users] Caller ID questions   Ryan Wagoner
• [asterisk-users] Working with Blacklist database   Myles Wakeham
• [asterisk-users] Manipulating the Blacklist database   Myles Wakeham
• [asterisk-users] Checking blank CallerID in Dialplan   Myles Wakeham
• [asterisk-users] REALTIME in 1.2   Jason Walker
• [asterisk-users] pattern containing an asterisk   C. Chad Wallace
• [asterisk-users] Productivity Suite on Polycom IP7000   Watkins, Bradley
• [asterisk-users] Registering a Cisco 7965 on 1.4.26   David White
• [asterisk-users] OT: NAT in SPA922   David White
• [asterisk-users] Simulating a commercial SIP provider   Jaap Winius
• [asterisk-users] Simulating a commercial SIP provider   Jaap Winius
• [asterisk-users] Simulating a commercial SIP provider   Jaap Winius
• [asterisk-users] ISDN config: LBO values   Jaap Winius
• [asterisk-users] ISDN config: LBO values   Jaap Winius
• [asterisk-users] ISDN config: LBO values   Jaap Winius
• [asterisk-users] client-server encryption   Hans Witvliet
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Hans Witvliet
• [asterisk-users] Connect mobile to asterisk   Hans Witvliet
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Greg Woods
• [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?   Greg Woods
• [asterisk-users] AGI <==> DeadAGI   CHEN XUEQIN
• [asterisk-users] IAX2 - providers discontinuing support   Zeeshan Zakaria
• [asterisk-users] Continuing after a TIMEOUT(absolute)   Zeeshan Zakaria
• [asterisk-users] Normalizing called numbers   Zeeshan Zakaria
• [asterisk-users] asterisk-users Digest, Vol 70, Issue 63   Zeeshan Zakaria
• [asterisk-users] Best way to limit outgoing calls per trunk   Zeeshan Zakaria
• [asterisk-users] Why Manager account log on and log off alternatively all the time?   Zeeshan Zakaria
• [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality   Zoa
• [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality   Zoa
• [asterisk-users] Being attacked by an Amazon EC2 ...   --[ UxBoD ]--
• [asterisk-users] Interesting email project.   --[ UxBoD ]--
• [asterisk-users] is my PHPAGI Soap code right?   --[ UxBoD ]--
• [asterisk-users] is my PHPAGI Soap code right?   --[ UxBoD ]--
• [asterisk-users] DAHDI Help (made a cardinal sin :()   --[ UxBoD ]--
• [asterisk-users] DAHDI Help (made a cardinal sin :()   --[ UxBoD ]--
• [asterisk-users] DAHDI Help (made a cardinal sin :()   --[ UxBoD ]--
• [asterisk-users] DAHDI Help (made a cardinal sin :()   --[ UxBoD ]--
• [asterisk-users] DAHDI Help (made a cardinal sin :()   --[ UxBoD ]--
• [asterisk-users] voipmonitor.org   mosbah abdelkader
• [asterisk-users] voipmonitor.org   mosbah abdelkader
• [asterisk-users] voipmonitor.org   mosbah abdelkader
• [asterisk-users] Customizing Asterisknow distribution   mosbah abdelkader
• [asterisk-users] About testing Call transfer in asterisk   mosbah abdelkader
• [asterisk-users] Sip phone does not call   ayodele abejide
• [asterisk-users] Installing sounds   ayodele abejide
• [asterisk-users] Implementing Asterisk on a WAN or an Intranet   ayodele abejide
• [asterisk-users] a2billing DID and Queues   toqeer ali
• [asterisk-users] bug in asterisk   benoit bernard
• [asterisk-users] Problems with Asterisk 1.6.2.1 working in Realtime with PostgreSQL   Renato bianchini
• [asterisk-users] Manual Web-meetme   Renato bianchini
• [asterisk-users] Problem of "Playing 'pbx-transfer'"   kamrun nahar bina
• [asterisk-users] Problem of hearing attended transfer' s sound   kamrun nahar bina
• [asterisk-users] Problem of hearing transfer' s sound　   kamrun nahar bina
• [asterisk-users] Problem of "Playing 'pbx-transfer'"   kamrun nahar bina
• [asterisk-users] Problem of "Playing 'pbx-transfer'"   kamrun nahar bina
• [asterisk-users] problem of "Cannot release Channel"   kamrun nahar bina
• [asterisk-users] Channel cannot be released   kamrun nahar bina
• [asterisk-users] Sangoma A101D PRI failing with ERROR - -- Got SABME from network peer. Sending Unnumbered Acknowledgement   bruce bruce
• [asterisk-users] Do you think my server is being attacked?   bruce bruce
• [asterisk-users] aastra pt 480e phone   bruce bruce
• [asterisk-users] q931.c modifications for CLID Presentation   bruce bruce
• [asterisk-users] Re-compiling q931.c   bruce bruce
• [asterisk-users] Re-compiling q931.c   bruce bruce
• [asterisk-users] OK, I'm stumped   bruce bruce
• [asterisk-users] PRI down due to chan_zap.c: No more room in scheduler....Got SABME and Sending Unnumbered Acknowledgement...Any thoughts?   bruce bruce
• [asterisk-users] Commands to upgrade to latest Libpri? can I upgrade without touching Asterisk?   bruce bruce
• [asterisk-users] Commands to upgrade to latest Libpri? can I upgrade without touching Asterisk?   bruce bruce
• [asterisk-users] Commands to upgrade to latest Libpri? can I upgrade without touching Asterisk?   bruce bruce
• [asterisk-users] Asterisk and RFC 3261   bruce bruce
• [asterisk-users] Softphones on thin clients...   bruce bruce
• [asterisk-users] Libpri 1.4.11 Released   bruce bruce
• [asterisk-users] q931.c modifications for CLID Presentation   bruce bruce
• [asterisk-users] "pri show version" still shows old version despite doing a make && make clean && make install for v1.4.11   bruce bruce
• [asterisk-users] "pri show version" still shows old version despite doing a make && make clean && make install for v1.4.11   bruce bruce
• [asterisk-users] "pri show version" still shows old version despite doing a make && make clean && make install for v1.4.11   bruce bruce
• [asterisk-users] Best way to limit outgoing calls per trunk   bruce bruce
• [asterisk-users] Best way to limit outgoing calls per trunk   bruce bruce
• [asterisk-users] Best way to limit outgoing calls per trunk   bruce bruce
• [asterisk-users] Best way to limit outgoing calls per trunk   bruce bruce
• [asterisk-users] Best way to limit outgoing calls per trunk   bruce bruce
• [asterisk-users] Best way to limit outgoing calls per trunk   bruce bruce
• [asterisk-users] aastra pt 480e phone   michael capelle
• [asterisk-users] Dahdi problems with kernel 2.6.32   covici at ccs.covici.com
• [asterisk-users] Dahdi problems with kernel 2.6.32   covici at ccs.covici.com
• [asterisk-users] Jack in /usr/local/ means failure for asterisk   covici at ccs.covici.com
• [asterisk-users] Dahdi problems with kernel 2.6.32   covici at ccs.covici.com
• [asterisk-users] Asterisk T.38 Gateway code testing   marek cervenka
• [asterisk-users] RTP ports   voip crazy
• [asterisk-users] problem with trustrpid   crjw
• [asterisk-users] file command with alaw file   crjw
• [asterisk-users] working example of t38 fax w/ 1.6.2?   sean darcy
• [asterisk-users] working example of t38 fax w/ 1.6.2?   sean darcy
• [asterisk-users] Asterisk 1.6.2.7 Now Available   sean darcy
• [asterisk-users] working example of t38 fax w/ 1.6.2?   sean darcy
• [asterisk-users] Still true: only first peer matched on incoming call?   sean darcy
• [asterisk-users] working example of t38 fax w/ 1.6.2?   sean darcy
• [asterisk-users] 1.6.2.8: need sip reload to reach peers.   sean darcy
• [asterisk-users] Channels In Use   dotnetdub
• [asterisk-users] Channels In Use   dotnetdub
• [asterisk-users] routing of calls   salaheddine elharit
• [asterisk-users] routing of calls   salaheddine elharit
• [asterisk-users] routing of calls   salaheddine elharit
• [asterisk-users] client-server encryption   iscario at free.fr
• [asterisk-users] DTMF Input from the User   taimur hasan
• [asterisk-users] DTMF Input from the User   taimur hasan
• [asterisk-users] Music on Hold   taimur hasan
• [asterisk-users] Music on Hold   taimur hasan
• [asterisk-users] US "Truth in caller id act"... and it's impact onservices   joea at j4computers.com
• [asterisk-users] asterisk-users Digest, Vol 70, Issue 63   vijay kumar
• [asterisk-users] Transfer calls using ##   hin lee
• [asterisk-users] Transfer calls using ##   hin lee
• [asterisk-users] Digium TE121P + DAHDI   hin lee
• [asterisk-users] Continuing after a TIMEOUT(absolute)   lesouvage
• [asterisk-users] Connect mobile to asterisk   lesouvage
• [asterisk-users] call-waiting   bhrugu mehta
• [asterisk-users] Interesting email project.   mike mosier
• [asterisk-users] Interesting email project.   mike mosier
• [asterisk-users] Interesting email project.   mike mosier
• [asterisk-users] Re TrixBox   mike mosier
• [asterisk-users] AsteriskNow   mike mosier
• [asterisk-users] Speech/DTMF mix?   mike mosier
• [asterisk-users] Aastra SIP phone regisration problems   mike mosier
• [asterisk-users] Problems with Asterisk and two Linksys SPA941   mike mosier
• [asterisk-users] Softphones on thin clients...   mgraves at mstvp.com
• [asterisk-users] State of JACK support i9n Asterisk   mgraves at mstvp.com
• [asterisk-users] meetme and jitterbuffer   nakaji
• [asterisk-users] problem with ringinuse=no, queue members receive randomly two calls   nik600
• [asterisk-users] problem with ringinuse=no, queue members receive randomly two calls   nik600
• [asterisk-users] problem with ringinuse=no, queue members receive randomly two calls   nik600
• [asterisk-users] Re TrixBox   ram
• [asterisk-users] voipmonitor.org   ram
• [asterisk-users] Manual Web-meetme   ram
• [asterisk-users] Asterisk Query   garge rama
• [asterisk-users] Asterisk Query   garge rama
• [asterisk-users] CallerID problem with astribank   frangky robert
• [asterisk-users] problems with unicall   Marcelo nunes dos santos
• [asterisk-users] Location with PRI / Analog lines   mir shahnawaz
• [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 mailin another PBX ?!   khalid touati
• [asterisk-users] Code in extensions.conf to leave a voicemailin another PBX ?!   khalid touati
• [asterisk-users] Faxes from website works, but from regular don't: cause 16   khalid touati
• [asterisk-users] Dead air between answer and packet2packet bridge (Bug 12708?)   tylmann
• [asterisk-users] iax calls via checkbox.cc   lists-asterisk-users at yoinks.net
• [asterisk-users] iax calls via checkbox.cc   lists-asterisk-users at yoinks.net

Last message date: Mon May 31 21:29:23 CDT 2010
Archived on: Mon May 31 21:23:05 CDT 2010

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