January 2012 Archives by author

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Starting: Sun Jan 1 06:22:50 CST 2012
Ending: Tue Jan 31 19:12:06 CST 2012
Messages: 902

• [asterisk-users] Mark queue agent as away   Raj Mathur ( राज माथुर )
• [asterisk-users] best softphone for 2012?   Raj Mathur ( राज माथुर )
• [asterisk-users] Queue member is permanently BUSY   Raj Mathur ( राज माथुर )
• [asterisk-users] Timing Slips CRC & E-Bit Errors - Asterisk - Trixbox 2.8.0.4   Liban Abdi
• [asterisk-users] configuring a Cisco 7961 so that different line appearances register to different SIP proxy addresses   Ken Alker
• [asterisk-users] ssh to a Cisco 7961 is not working   Ken Alker
• [asterisk-users] ssh to a Cisco 7961 is not working   Ken Alker
• [asterisk-users] ssh to a Cisco 7961 is not working   Ken Alker
• [asterisk-users] Best non polycom SIP conference room phone   Carlos Alvarez
• [asterisk-users] Best non polycom SIP conference room phone   Carlos Alvarez
• [asterisk-users] Best non polycom SIP conference room phone   Carlos Alvarez
• [asterisk-users] Best non polycom SIP conference room phone   Carlos Alvarez
• [asterisk-users] Hang up phone after declined attended transfer   Carlos Alvarez
• [asterisk-users] Hang up phone after declined attended transfer   Carlos Alvarez
• [asterisk-users] Hang up phone after declined attended transfer   Carlos Alvarez
• [asterisk-users] SIP hardphone with dual gigabit ethernet ports   Carlos Alvarez
• [asterisk-users] Cordless SIP phone   Carlos Alvarez
• [asterisk-users] Cordless SIP phone   Carlos Alvarez
• [asterisk-users] Cordless SIP phone   Carlos Alvarez
• [asterisk-users] Anyone have a reliable T.38 Solution   Benny Amorsen
• [asterisk-users] Best non polycom SIP conference room phone   Benny Amorsen
• [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?   Benny Amorsen
• [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?   Benny Amorsen
• [asterisk-users] Cordless SIP phone   Benny Amorsen
• [asterisk-users] Asterisk 10.0 & 1.4 - iax codec are not compatible   Andres
• [asterisk-users] Q: SIPNATtraversal.pdf   Matthias Apitz
• [asterisk-users] AMI - Getting Event of QueueAgents WrapupTime State   Karsten Asche
• [asterisk-users] Chan_Mobile Nokia E51, csr bt dogle, Voice OK but no SMS Support ?   Din Assegaf
• [asterisk-users] process_sdp: Unsupported SDP media type in offer: audio , Failing due to no acceptable offer found   Din Assegaf
• [asterisk-users] process_sdp: Unsupported SDP media type in offer: audio , Failing due to no acceptable offer found   Din Assegaf
• [asterisk-users] Problems with codec translation when using Monitor and MixMonitor   Daniel - Asterisk
• [asterisk-users] Problems with codec translation when using Monitor and MixMonitor   Daniel - Asterisk
• [asterisk-users] Problems with codec translation when using Monitor and MixMonitor   Daniel - Asterisk
• [asterisk-users] Wired attack on Asterisk - Can anyone explain this?   Bruce B
• [asterisk-users] Speech recognition in asterisk using google voice API   Bruce B
• [asterisk-users] Speech recognition in asterisk using google voice API   Bruce B
• [asterisk-users] Speech recognition in asterisk using google voice API   Bruce B
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Bruce B
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Bruce B
• [asterisk-users] Speech recognition in asterisk using google voice API   Bruce B
• [asterisk-users] Speech recognition in asterisk using google voice API   Bruce B
• [asterisk-users] Speech recognition in asterisk using google voice API   Bruce B
• [asterisk-users] Speech recognition in asterisk using google voice API   Bruce B
• [asterisk-users] Anyone have a reliable T.38 Solution   David Backeberg
• [asterisk-users] DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0 Released   David Backeberg
• [asterisk-users] Odd DTMF problem when receiving calls   David Backeberg
• [asterisk-users] Executing Script after MixMonitor is called   David Backeberg
• [asterisk-users] Executing Script after MixMonitor is called   David Backeberg
• [asterisk-users] Executing Script after MixMonitor is called   David Backeberg
• [asterisk-users] Cisco AS5300 and Digium g729A codec   Alex Balashov
• [asterisk-users] SDP Issue   Alex Balashov
• [asterisk-users] SDP Issue   Alex Balashov
• [asterisk-users] SDP Issue   Alex Balashov
• [asterisk-users] SDP Issue   Alex Balashov
• [asterisk-users] Executing Script after MixMonitor is called   Satish Barot
• [asterisk-users] Asterisk1.8 support video trancoding ?   Paul Belanger
• [asterisk-users] Core file created in /tmp   Paul Belanger
• [asterisk-users] Compile error 1.8.8.1   Paul Belanger
• [asterisk-users] upgraded 1.8.8.0 > 10.1.0-rc2: now db warnings   Paul Belanger
• [asterisk-users] Pickup calls coming from queues   Niccolò Belli
• [asterisk-users] Pickup calls coming from queues   Niccolò Belli
• [asterisk-users] Pickup calls coming from queues   Niccolò Belli
• [asterisk-users] Pickup calls coming from queues   Niccolò Belli
• [asterisk-users] Cell Phone as a Queue member   Niccolò Belli
• [asterisk-users] Cell Phone as a Queue member   Niccolò Belli
• [asterisk-users] Cell Phone as a Queue member   Niccolò Belli
• [asterisk-users] Proposed changes to Asterisk release and support cycles   Niccolò Belli
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)   Ron Bergin
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus(oracle)   Ron Bergin
• [asterisk-users] question sangoma vs digium   Agustina Berretta
• [asterisk-users] message WARNING[] features.c: Failed to play transfer sound! and attended transfer hangs up   Agustina Berretta
• [asterisk-users] Huh? Local is being asked to answer?   Agustina Berretta
• [asterisk-users] Softphones with SIP transfer   Agustina Berretta
• [asterisk-users] Manager Originate and Callerid ?   Russell Brown
• [asterisk-users] best softphone for 2012?   Ross Cameron
• [asterisk-users] Real trunk group w/ DAHDI   Louis Carreiro
• [asterisk-users] Real trunk group w/ DAHDI   Louis Carreiro
• [asterisk-users] Real T1 trunk group...   Louis Carreiro
• [asterisk-users] Real T1 trunk group...   Louis Carreiro
• [asterisk-users] Real T1 trunk group...   Louis Carreiro
• [asterisk-users] Server-to-server BLF   Ronald Cepres
• [asterisk-users] Asterisk as register server through OpenSIPS   Ronald Cepres
• [asterisk-users] Server-to-server BLF   Ronald Cepres
• [asterisk-users] Server-to-server BLF   Ronald Cepres
• [asterisk-users] Avaya 4610sw IP Phone   Aamir Chougule
• [asterisk-users] Avaya 4610sw IP Phone   Aamir Chougule
• [asterisk-users] Installing the 3.1 sources of Kernel with Asterisk   Christian
• [asterisk-users] Does Asterisk permit multiple registrations to the same host?   Frank Church
• [asterisk-users] Does Asterisk permit multiple registrations to the same host?   Frank Church
• [asterisk-users] Asterisk won't start - trap invalid opcode   James Cloos
• [asterisk-users] cached VMI on manual voicemail update   Tzafrir Cohen
• [asterisk-users] create table in mysql using asterisk   Tzafrir Cohen
• [asterisk-users] create table in mysql using asterisk   Tzafrir Cohen
• [asterisk-users] cached VMI on manual voicemail update   Tzafrir Cohen
• [asterisk-users] How to check currently used libraries from command line ?   Tzafrir Cohen
• [asterisk-users] Starting things off without a dial tone   Tzafrir Cohen
• [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?   Tzafrir Cohen
• [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?   Tzafrir Cohen
• [asterisk-users] Installing the 3.1 sources of Kernel with Asterisk   Tzafrir Cohen
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Tzafrir Cohen
• [asterisk-users] Efficient logging of PRI traffic for later analysis?   Tzafrir Cohen
• [asterisk-users] best softphone for 2012?   Sean Darcy
• [asterisk-users] Server-to-server BLF   Leandro Dardini
• [asterisk-users] Does Asterisk permit multiple registrations to the same host?   Leandro Dardini
• [asterisk-users] Anyone have a reliable T.38 Solution   Matt Darnell
• [asterisk-users] Anyone have a reliable T.38 Solution   Matt Darnell
• [asterisk-users] SIP and NAT best practices since recent changes?   Steve Davies
• [asterisk-users] SIP and NAT best practices since recent changes?   Steve Davies
• [asterisk-users] asterisk problem sip   Alec Davis
• [asterisk-users] Pickup calls coming from queues   Alec Davis
• [asterisk-users] Pickup calls coming from queues   Alec Davis
• [asterisk-users] Pickup calls coming from queues   Alec Davis
• [asterisk-users] Strange how Asterisk know the updated information of log   Alec Davis
• [asterisk-users] TCP transport and BLF   Alec Davis
• [asterisk-users] SDP Issue   Phil Daws
• [asterisk-users] SDP Issue   Phil Daws
• [asterisk-users] RFE idea for VM application   Phil Daws
• [asterisk-users] RFE idea for VM application   Phil Daws
• [asterisk-users] RFE idea for VM application   Phil Daws
• [asterisk-users] RFE idea for VM application   Phil Daws
• [asterisk-users] RFE idea for VM application   Phil Daws
• [asterisk-users] Problem w/ PC port on Polycom 335   Jim DeVito
• [asterisk-users] Question on system command 1.4.43   Jim DeVito
• [asterisk-users] Best non polycom SIP conference room phone   Jim DeVito
• [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?   Jim DeVito
• [asterisk-users] Call abandoned from queue not showing in CDR (possible bug)   Jim DeVito
• [asterisk-users] Sip Registration Hijacking   Jim DeVito
• [asterisk-users] Force CDR to be written.   Jim DeVito
• [asterisk-users] allowguest = yes? no?   Jim DeVito
• [asterisk-users] upgraded 1.8.8.0 > 10.1.0-rc2: now db warnings   Jim DeVito
• [asterisk-users] Asterisk to support Dialogic Cards   Vinod Dharashive
• [asterisk-users] Asterisk to support Dialogic Cards   Vinod Dharashive
• [asterisk-users] Too many open files   Juan David Diaz
• [asterisk-users] calling specific 1800-number not going through.   Jim Dickenson
• [asterisk-users] Answering call from queue, then put back in queue?   Jim Dickenson
• [asterisk-users] Problems with codec translation when using Monitor and MixMonitor   Jim Dickenson
• [asterisk-users] Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1   Jim Dickenson
• [asterisk-users] Problem w/ PC port on Polycom 335   Mike Diehl
• [asterisk-users] Problem w/ PC port on Polycom 335   Mike Diehl
• [asterisk-users] Problem answering phone   Mike Diehl
• [asterisk-users] Problem answering phone   Mike Diehl
• [asterisk-users] Too many open files   Mike Diehl
• [asterisk-users] SendFax not sending AMI events   Mike Diehl
• [asterisk-users] Too many open files   Mike Diehl
• [asterisk-users] SendFax not sending AMI events   Mike Diehl
• [asterisk-users] Preços por serviços e equipamento   Cláudio Duarte
• [asterisk-users] Speech recognition in asterisk using google voice API   Michelle Dupuis
• [asterisk-users] View # active calls in a context   Michelle Dupuis
• [asterisk-users] Question on system command 1.4.43   Steve Edwards
• [asterisk-users] Asterisk won't start - trap invalid opcode   Steve Edwards
• [asterisk-users] Rami   Steve Edwards
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)   Steve Edwards
• [asterisk-users] STOP loading extensions.ael   Steve Edwards
• [asterisk-users] Why write your dialplan using Lua?   Steve Edwards
• [asterisk-users] Why write your dialplan using Lua?   Steve Edwards
• [asterisk-users] Why write your dialplan using Lua?   Steve Edwards
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Steve Edwards
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus(oracle)   Steve Edwards
• [asterisk-users] create table in mysql using asterisk   Steve Edwards
• [asterisk-users] /etc/init.d script and calling asterisk command line.   Steve Edwards
• [asterisk-users] Sip Registration Hijacking   Steve Edwards
• [asterisk-users] Executing Script after MixMonitor is called   Steve Edwards
• [asterisk-users] CA Issued Certificates / TLS + SRTP   Stuart Elvish
• [asterisk-users] CA Issued Certificates / TLS + SRTP   Stuart Elvish
• [asterisk-users] Avaya 4610sw IP Phone   Shaun Ewing
• [asterisk-users] create table in mysql using asterisk   Eyal
• [asterisk-users] create table in mysql using asterisk   Eyal
• [asterisk-users] Change the caller's phone number   Eyal
• [asterisk-users] Change the caller's phone number   Eyal
• [asterisk-users] play sound file   Eyal
• [asterisk-users] play sound file   Eyal
• [asterisk-users] play sound file   Eyal
• [asterisk-users] Best non polycom SIP conference room phone   C F
• [asterisk-users] Connecting to an Old Phone System   C F
• [asterisk-users] Best non polycom SIP conference room phone   C F
• [asterisk-users] Best non polycom SIP conference room phone   C F
• [asterisk-users] Best non polycom SIP conference room phone   C F
• [asterisk-users] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?   C F
• [asterisk-users] Real T1 trunk group...   C F
• [asterisk-users] Cell Phone as a Queue member   C F
• [asterisk-users] asterisk 1.8 codec negotiation   Kevin P. Fleming
• [asterisk-users] NAT/IPTABLES workarounds   Kevin P. Fleming
• [asterisk-users] Anyone have a reliable T.38 Solution   Kevin P. Fleming
• [asterisk-users] Video trancoding not done.   Kevin P. Fleming
• [asterisk-users] Asterisk1.8 support video trancoding ?   Kevin P. Fleming
• [asterisk-users] Where are the fax instructions?   Kevin P. Fleming
• [asterisk-users] asterisk 1.8 codec negotiation   Kevin P. Fleming
• [asterisk-users] Best non polycom SIP conference room phone   Kevin P. Fleming
• [asterisk-users] Where are the fax instructions?   Kevin P. Fleming
• [asterisk-users] Where are the fax instructions?   Kevin P. Fleming
• [asterisk-users] 44Khz files in Asterisk 10   Kevin P. Fleming
• [asterisk-users] Anyone have a reliable T.38 Solution   Kevin P. Fleming
• [asterisk-users] SIP and NAT best practices since recent changes?   Kevin P. Fleming
• [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED]   Kevin P. Fleming
• [asterisk-users] SIP and NAT best practices since recent changes?   Kevin P. Fleming
• [asterisk-users] ConfBridge no audio problem   Kevin P. Fleming
• [asterisk-users] Questions on hardware or software-based echo cancellation   Kevin P. Fleming
• [asterisk-users] Problems with codec translation when using Monitor and MixMonitor   Kevin P. Fleming
• [asterisk-users] Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1   Kevin P. Fleming
• [asterisk-users] Questions on hardware or software-based echo cancellation   Kevin P. Fleming
• [asterisk-users] CDR into ical?   Kevin P. Fleming
• [asterisk-users] Update callee num or name at caller display   Kevin P. Fleming
• [asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?   Kevin P. Fleming
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Kevin P. Fleming
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Kevin P. Fleming
• [asterisk-users] Failed to Allocate port for RTP instance   Kevin P. Fleming
• [asterisk-users] Real trunk group w/ DAHDI   Kevin P. Fleming
• [asterisk-users] Asterisk to support Dialogic Cards   Kevin P. Fleming
• [asterisk-users] Asterisk rewrites "From" header when CALLERID(num-pres)=prohib_passed_screen is set   Kevin P. Fleming
• [asterisk-users] Asterisk to support Dialogic Cards   Kevin P. Fleming
• [asterisk-users] Asterisk to support Dialogic Cards   Kevin P. Fleming
• [asterisk-users] Efficient logging of PRI traffic for later analysis?   Kevin P. Fleming
• [asterisk-users] Asterisk NOT in the media path   Kevin P. Fleming
• [asterisk-users] Pickup calls coming from queues   Kevin P. Fleming
• [asterisk-users] SDP Issue   Kevin P. Fleming
• [asterisk-users] allowguest = yes? no?   Kevin P. Fleming
• [asterisk-users] ConfBridge details   Kevin P. Fleming
• [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality   Kevin P. Fleming
• [asterisk-users] User hit f to disconnect call.   Kevin P. Fleming
• [asterisk-users] Manager Originate and Callerid ?   Kevin P. Fleming
• [asterisk-users] process_sdp: Unsupported SDP media type in offer: audio , Failing due to no acceptable offer found   Kevin P. Fleming
• [asterisk-users] SendFax not sending AMI events   Kevin P. Fleming
• [asterisk-users] Problem with DTMF in Voicemail main   Kevin P. Fleming
• [asterisk-users] Proposed changes to Asterisk release and support cycles   Kevin P. Fleming
• [asterisk-users] Proposed changes to Asterisk release and support cycles   Kevin P. Fleming
• [asterisk-users] Proposed changes to Asterisk release and support cycles   Kevin P. Fleming
• [asterisk-users] Proposed changes to Asterisk release and support cycles   Kevin P. Fleming
• [asterisk-users] Troubleshooting one-way audio with H.323 trunk between Asterisk and Avaya IP Office   Phil Frost
• [asterisk-users] Question on system command 1.4.43   Jerry Geis
• [asterisk-users] Question on system command 1.4.43   Jerry Geis
• [asterisk-users] question on CDR   Jerry Geis
• [asterisk-users] Asterisk 1.8.9.0 Now Available   Eric Germann
• [asterisk-users] Asterisk 1.8.9.0 Now Available   Eric Germann
• [asterisk-users] Couple of questions: SIP ALG, allowguest=no   Gilles
• [asterisk-users] best softphone for 2012?   Gilles
• [asterisk-users] best softphone for 2012?   Gilles
• [asterisk-users] allowguest = yes? no?   Gilles
• [asterisk-users] allowguest = yes? no?   Gilles
• [asterisk-users] allowguest = yes? no?   Gilles
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Gilles
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Gilles
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Gilles
• [asterisk-users] SRV record for non-standard SIP port?   Gilles
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Gilles
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Gilles
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Gilles
• [asterisk-users] SRV record for non-standard SIP port?   Gilles
• [asterisk-users] Speech recognition in asterisk using google voice API   Israel Gottlieb
• [asterisk-users] Set Call type in dial plan   Sammy Govind
• [asterisk-users] Set Call type in dial plan   Sammy Govind
• [asterisk-users] Using Asterisk as a softphone   Sammy Govind
• [asterisk-users] From address missing 'sip:', using it anyway   Sammy Govind
• [asterisk-users] NAT/IPTABLES workarounds   Sammy Govind
• [asterisk-users] Exceptionally long voice queue length   Sammy Govind
• [asterisk-users] Asterisk as UAC: How to put call OnHold   Sammy Govind
• [asterisk-users] Asterisk as UAC: How to put call OnHold   Sammy Govind
• [asterisk-users] Peer doesn't answer   Sammy Govind
• [asterisk-users] Peer doesn't answer   Sammy Govind
• [asterisk-users] Macro vs sub   Sammy Govind
• [asterisk-users] Macro vs sub   Sammy Govind
• [asterisk-users] SDP Issue   Sammy Govind
• [asterisk-users] play sound file   Sammy Govind
• [asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller   Luke Hamburg
• [asterisk-users] Best non polycom SIP conference room phone   Luke Hamburg
• [asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller   Luke Hamburg
• [asterisk-users] registration not authorized - stale nonce   Matt Hamilton
• [asterisk-users] UPDATE RE: registration not authorized - stale nonce   Matt Hamilton
• [asterisk-users] public ip issue with asterisk cluster   Matt Hamilton
• [asterisk-users] meetme - Unable to write frame to channel   Matt Hamilton
• [asterisk-users] meetme - Unable to write frame to channel   Matt Hamilton
• [asterisk-users] meetme - Unable to write frame to channel   Matt Hamilton
• [asterisk-users] AstLinux 1.01 Released   Darrick Hartman
• [asterisk-users] Compile error 1.8.8.1   Nyamul Hassan
• [asterisk-users] Compile error 1.8.8.1   Nyamul Hassan
• [asterisk-users] Connecting to an Old Phone System   Paul Hayes
• [asterisk-users] Change port from 5060 on Snom phone   Paul Hayes
• [asterisk-users] local channels and g729a voice quality   Paul Hayes
• [asterisk-users] Sip Registration Hijacking   Paul Hayes
• [asterisk-users] Odd DTMF problem when receiving calls   Christopher David Howie
• [asterisk-users] Odd DTMF problem when receiving calls   Christopher David Howie
• [asterisk-users] Call abandoned from queue not showing in CDR (possible bug)   SIP IMS
• [asterisk-users] Signalling and Media Configuration   Ryan Icasiano
• [asterisk-users] Sip Registration Hijacking   Alejandro Imass
• [asterisk-users] Sip Registration Hijacking   Alejandro Imass
• [asterisk-users] FAX Installation in Asterisk   Ioan Indreias
• [asterisk-users] play sound file   Nasir Iqbal
• [asterisk-users] Cordless SIP phone   Ira
• [asterisk-users] Problem with DTMF in Voicemail main   Ira
• [asterisk-users] Using Asterisk as a softphone   Christian Jaeger
• [asterisk-users] Asterisk as register server through OpenSIPS   Olle E. Johansson
• [asterisk-users] odbc storage for video message   Matthew Jordan
• [asterisk-users] Macro vs sub   Matthew Jordan
• [asterisk-users] STOP loading extensions.ael   Joseph
• [asterisk-users] STOP loading extensions.ael   Joseph
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Joseph
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Joseph
• [asterisk-users] calling specific 1800-number not going through.   Joseph
• [asterisk-users] calling specific 1800-number not going through.   Joseph
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Joseph
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Joseph
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Joseph
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Joseph
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Joseph
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Joseph
• [asterisk-users] Asterisk 10.0 & 1.4 - iax codec are not compatible   Joseph
• [asterisk-users] Asterisk 10.0 & 1.4 - iax codec are not compatible   Joseph
• [asterisk-users] [SOLVED] Asterisk 10.0 & 1.4 - iax codec are not compatible   Joseph
• [asterisk-users] [SOLVED] Asterisk 10.0 & 1.4 - iax codec are not compatible   Joseph
• [asterisk-users] [SOLVED - need explanation] Asterisk 10.0 & 1.4 - iax codec are not compatible   Joseph
• [asterisk-users] Asterisk 1.8 - SIP losing registration   Joseph
• [asterisk-users] SIP - connected line has changed. Saving it until answer for IAX2/iaxy   Joseph
• [asterisk-users] Connecting to an Old Phone System   Dan Journo
• [asterisk-users] dialplan problem : not including context   Jonas Kellens
• [asterisk-users] dialplan problem : not including context   Jonas Kellens
• [asterisk-users] dialplan problem : not including context   Jonas Kellens
• [asterisk-users] dialplan problem : not including context   Jonas Kellens
• [asterisk-users] dialplan problem : not including context   Jonas Kellens
• [asterisk-users] dialplan problem : not including context   Jonas Kellens
• [asterisk-users] Core file created in /tmp   Jonas Kellens
• [asterisk-users] Macro vs sub   Jonas Kellens
• [asterisk-users] Core file created in /tmp   Jonas Kellens
• [asterisk-users] Macro vs sub   Jonas Kellens
• [asterisk-users] Macro vs sub   Jonas Kellens
• [asterisk-users] Macro vs sub   Jonas Kellens
• [asterisk-users] Asterisk NOT in the media path   Jonas Kellens
• [asterisk-users] Asterisk NOT in the media path   Jonas Kellens
• [asterisk-users] ChanSpy : how to know channel name ?   Jonas Kellens
• [asterisk-users] ChanSpy : how to know channel name ?   Jonas Kellens
• [asterisk-users] ChanSpy : how to know channel name ?   Jonas Kellens
• [asterisk-users] ChanSpy : how to know channel name ?   Jonas Kellens
• [asterisk-users] ChanSpy : how to know channel name ?   Jonas Kellens
• [asterisk-users] ChanSpy : how to know channel name ?   Jonas Kellens
• [asterisk-users] ChanSpy : how to know channel name ?   Jonas Kellens
• [asterisk-users] dialplan problem : not including context   Jonas Kellens
• [asterisk-users] ChanSpy : how to know channel name ?   Jonas Kellens
• [asterisk-users] ChanSpy : how to know channel name ?   Jonas Kellens
• [asterisk-users] Connecting to an Old Phone System   Don Kelly
• [asterisk-users] Anyone have a reliable T.38 Solution   Michael Keuter
• [asterisk-users] Pickup calls coming from queues   Michael Keuter
• [asterisk-users] Pickup calls coming from queues   Michael Keuter
• [asterisk-users] Set Call type in dial plan   Faraj Khasib
• [asterisk-users] Set Call type in dial plan   Faraj Khasib
• [asterisk-users] Set Call type in dial plan   Faraj Khasib
• [asterisk-users] Set Call type in dial plan   Faraj Khasib
• [asterisk-users] Set Call type in dial plan   Faraj Khasib
• [asterisk-users] Set Call type in dial plan   Faraj Khasib
• [asterisk-users] Set Call type in dial plan   Faraj Khasib
• [asterisk-users] How to make SIP guest call   Faraj Khasib
• [asterisk-users] How to make SIP guest call   Faraj Khasib
• [asterisk-users] How to make SIP guest call   Faraj Khasib
• [asterisk-users] How to make SIP guest call   Faraj Khasib
• [asterisk-users] How to make SIP guest call   Faraj Khasib
• [asterisk-users] Registering multi-clients   Faraj Khasib
• [asterisk-users] How to make SIP guest call   Faraj Khasib
• [asterisk-users] How to make SIP guest call   Faraj Khasib
• [asterisk-users] Set Call Codec in extension.conf   Faraj Khasib
• [asterisk-users] Set Call Codec in extension.conf   Faraj Khasib
• [asterisk-users] Set Call Codec in extension.conf   Faraj Khasib
• [asterisk-users] Set Call Codec in extension.conf   Faraj Khasib
• [asterisk-users] Set Call Codec in extension.conf   Faraj Khasib
• [asterisk-users] Set Call Codec in extension.conf   Faraj Khasib
• [asterisk-users] Set Call Codec in extension.conf   Faraj Khasib
• [asterisk-users] Set Call Codec in extension.conf   Faraj Khasib
• [asterisk-users] Set Call Codec in extension.conf   Faraj Khasib
• [asterisk-users] Set Call Codec in extension.conf   Faraj Khasib
• [asterisk-users] Set Call Codec in extension.conf   Faraj Khasib
• [asterisk-users] Set Call Codec in extension.conf   Faraj Khasib
• [asterisk-users] Set Call type in dial plan   Faraj Khasib
• [asterisk-users] Change the caller's phone number   Faraj Khasib
• [asterisk-users] Change the caller's phone number   Faraj Khasib
• [asterisk-users] Executing Script after MixMonitor is called   Faraj Khasib
• [asterisk-users] Exceptionally long voice queue length   Vik Killa
• [asterisk-users] Exceptionally long voice queue length   Vik Killa
• [asterisk-users] Exceptionally long voice queue length   Vik Killa
• [asterisk-users] Exceptionally long voice queue length   Vik Killa
• [asterisk-users] ConfBridge details   Jeremy Kister
• [asterisk-users] ConfBridge details   Jeremy Kister
• [asterisk-users] ConfBridge details   Jeremy Kister
• [asterisk-users] Executing Script after MixMonitor is called   Jeremy Kister
• [asterisk-users] How to make SIP guest call   David Klaverstyn
• [asterisk-users] Anyone have a reliable T.38 Solution   David Klaverstyn
• [asterisk-users] Anyone have a reliable T.38 Solution   Klaverstyn, David C
• [asterisk-users] Dahdi for meetme on AMD64 arch?   John Knight
• [asterisk-users] Dahdi for meetme on AMD64 arch?   John Knight
• [asterisk-users] Dahdi for meetme on AMD64 arch?   John Knight
• [asterisk-users] Dahdi for meetme on AMD64 arch?   John Knight
• [asterisk-users] Proposed changes to Asterisk release and support cycles   John Knight
• [asterisk-users] Experience with Eicon Diva PRO 3.0?   Michelle Konzack
• [asterisk-users] DIALSTATUS Values   Kamlesh Kumar
• [asterisk-users] How to check currently used libraries from command line ?   Anton Kvashenkin
• [asterisk-users] Analoge and E1 ports   Anton Kvashenkin
• [asterisk-users] Speech recognition in asterisk using google voice API   LL
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)   LL
• [asterisk-users] odd disconnects with major company's voice recog   Jeff LaCoursiere
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Jeff LaCoursiere
• [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality   Jeff LaCoursiere
• [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality   Jeff LaCoursiere
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Jeff LaCoursiere
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Jeff LaCoursiere
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Jeff LaCoursiere
• [asterisk-users] Failed to authenticate on INVITE to Anonymous   Jayesh Labade
• [asterisk-users] Failed to authenticate on INVITE to Anonymous   Jayesh Labade
• [asterisk-users] Failed to authenticate on INVITE to Anonymous   Jayesh Labade
• [asterisk-users] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?   Alex Villací­s Lasso
• [asterisk-users] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?   Alex Villací­s Lasso
• [asterisk-users] Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1   Alex Villací­s Lasso
• [asterisk-users] Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1   Alex Villací­s Lasso
• [asterisk-users] Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1   Alex Villací­s Lasso
• [asterisk-users] Deadlock detected in asterisk-1.8.9.0 x86_64   Alex Villací­s Lasso
• [asterisk-users] Sip Registration Hijacking   Mikhail Lischuk
• [asterisk-users] Weird IPs in Fail2ban list   Mikhail Lischuk
• [asterisk-users] Asterisk 1.8 - BRI D Channel going up and down every few seconds   Patrick Lists
• [asterisk-users] How to make SIP guest call   Patrick Lists
• [asterisk-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.   Patrick Lists
• [asterisk-users] NAT/IPTABLES workarounds   Patrick Lists
• [asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?   Patrick Lists
• [asterisk-users] Problem answering phone   Patrick Lists
• [asterisk-users] local channels and g729a voice quality   Patrick Lists
• [asterisk-users] Asterisk to support Dialogic Cards   Patrick Lists
• [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality   Patrick Lists
• [asterisk-users] Experience with Eicon Diva PRO 3.0?   Patrick Lists
• [asterisk-users] Speech recognition in asterisk using google voice API   Julian Lyndon-Smith
• [asterisk-users] Speech recognition in asterisk using google voice API   Julian Lyndon-Smith
• [asterisk-users] Speech recognition in asterisk using google voice API   Julian Lyndon-Smith
• [asterisk-users] Set Call type in dial plan   Doug Lytle
• [asterisk-users] Problem w/ PC port on Polycom 335   Doug Lytle
• [asterisk-users] Question on system command 1.4.43   Doug Lytle
• [asterisk-users] dialplan problem : not including context   Doug Lytle
• [asterisk-users] dialplan problem : not including context   Doug Lytle
• [asterisk-users] dialplan problem : not including context   Doug Lytle
• [asterisk-users] dialplan problem : not including context   Doug Lytle
• [asterisk-users] dialplan problem : not including context   Doug Lytle
• [asterisk-users] dialplan problem : not including context   Doug Lytle
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Doug Lytle
• [asterisk-users] Voicemail weirdness after upgrade   Doug Lytle
• [asterisk-users] unsubscribe   Doug Lytle
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Doug Lytle
• [asterisk-users] Cell Phone as a Queue member   Doug Lytle
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Doug Lytle
• [asterisk-users] tcp version of toronto - osaka doesn't work   Leif Madsen
• [asterisk-users] Best non polycom SIP conference room phone   Leif Madsen
• [asterisk-users] Server-to-server BLF   Ishfaq Malik
• [asterisk-users] Change port from 5060 on Snom phone   Ishfaq Malik
• [asterisk-users] Best non polycom SIP conference room phone   Ishfaq Malik
• [asterisk-users] Call abandoned from queue not showing in CDR (possible bug)   Ishfaq Malik
• [asterisk-users] Call abandoned from queue not showing in CDR (possible bug)   Ishfaq Malik
• [asterisk-users] Pickup calls coming from queues   Ishfaq Malik
• [asterisk-users] Cordless SIP phone   Ishfaq Malik
• [asterisk-users] Cordless SIP phone   Ishfaq Malik
• [asterisk-users] ChanSpy : how to know channel name ?   Ishfaq Malik
• [asterisk-users] Blocking in: ast_waitfor_nandfds   Ishfaq Malik
• [asterisk-users] ChanSpy : how to know channel name ?   Ishfaq Malik
• [asterisk-users] SIP trunk call initiated as Anonymous at anonymous.invalid   Gordon Messmer
• [asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?   Russ Meyerriecks
• [asterisk-users] Timing Slips CRC & E-Bit Errors - Asterisk - Trixbox 2.8.0.4   Russ Meyerriecks
• [asterisk-users] ssh to a Cisco 7961 is not working   Vladimir Mikhelson
• [asterisk-users] AST-2012-001: SRTP Video Remote Crash Vulnerability   Vladimir Mikhelson
• [asterisk-users] Weird IPs in Fail2ban list   Barry Miller
• [asterisk-users] fail2ban restarts   Barry Miller
• [asterisk-users] ssh to a Cisco 7961 is not working   Flavio Miranda
• [asterisk-users] ssh to a Cisco 7961 is not working   Flavio Miranda
• [asterisk-users] Peer doesn't answer   Flavio Miranda
• [asterisk-users] Peer doesn't answer   Flavio Miranda
• [asterisk-users] Help_video voice mail not retriev properly   Durgesh Mishra
• [asterisk-users] Video trancoding not done.   Durgesh Mishra
• [asterisk-users] Asterisk1.8 support video trancoding ?   Durgesh Mishra
• [asterisk-users] video mail is not store   Durgesh Mishra
• [asterisk-users] Rami   Arjan Kroon | Mobillion
• [asterisk-users] Rami   Arjan Kroon | Mobillion
• [asterisk-users] Stuck DAHDI Channels   Antonio Modesto
• [asterisk-users] Asterisk 1.8 - BRI D Channel going up and down every few seconds   Marco Mooijekind
• [asterisk-users] Sip Registration Hijacking   Larry Moore
• [asterisk-users] /etc/init.d script and calling asterisk command line.   Luis Morales
• [asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller   Douglas Mortensen
• [asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller   Douglas Mortensen
• [asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller   Douglas Mortensen
• [asterisk-users] Asterisk 10.0 & 1.4 - iax codec are not compatible   Tony Mountifield
• [asterisk-users] create table in mysql using asterisk   Tony Mountifield
• [asterisk-users] create table in mysql using asterisk   Tony Mountifield
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Tony Mountifield
• [asterisk-users] Efficient logging of PRI traffic for later analysis?   Tony Mountifield
• [asterisk-users] Efficient logging of PRI traffic for later analysis?   Tony Mountifield
• [asterisk-users] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?   Richard Mudgett
• [asterisk-users] Noise in caller handset when dialing out (with dahdi 2.6.0)   Richard Mudgett
• [asterisk-users] message WARNING[] features.c: Failed to play transfer sound! and attended transfer hangs up   Richard Mudgett
• [asterisk-users] Where to find meaning of /n in Local/6613 at from-queue/n ?   Richard Mudgett
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)   Ahmed Munir
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)   Ahmed Munir
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus(oracle)   Ahmed Munir
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus(oracle)   Ahmed Munir
• [asterisk-users] 481 Call leg/transaction does not exists Status Response   Elliot Murdock
• [asterisk-users] 1.6 and 1.8   Elliot Murdock
• [asterisk-users] DEBUG Message   Elliot Murdock
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Elliot Murdock
• [asterisk-users] Peer doesn't answer   Arlen Nascimento
• [asterisk-users] Peer doesn't answer   Arlen Nascimento
• [asterisk-users] Peer doesn't answer   Arlen Nascimento
• [asterisk-users] Peer doesn't answer   Arlen Nascimento
• [asterisk-users] Peer doesn't answer   Arlen Nascimento
• [asterisk-users] Peer doesn't answer   Arlen Nascimento
• [asterisk-users] Asterisk 10.0 & 1.4 - iax codec are not compatible   Tim Nelson
• [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?   Tim Nelson
• [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality   Tim Nelson
• [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality   Tim Nelson
• [asterisk-users] NAT/IPTABLES workarounds   Danny Nicholas
• [asterisk-users] Set Call Codec in extension.conf   Danny Nicholas
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)   Danny Nicholas
• [asterisk-users] Set Call Codec in extension.conf   Danny Nicholas
• [asterisk-users] Set Call Codec in extension.conf   Danny Nicholas
• [asterisk-users] Set Call Codec in extension.conf   Danny Nicholas
• [asterisk-users] Set Call Codec in extension.conf   Danny Nicholas
• [asterisk-users] Set Call Codec in extension.conf   Danny Nicholas
• [asterisk-users] Set Call Codec in extension.conf   Danny Nicholas
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)   Danny Nicholas
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus(oracle)   Danny Nicholas
• [asterisk-users] question on CDR   Danny Nicholas
• [asterisk-users] STOP loading extensions.ael   Danny Nicholas
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus(oracle)   Danny Nicholas
• [asterisk-users] Problem connecting to 4569/UDP   Danny Nicholas
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Danny Nicholas
• [asterisk-users] Why write your dialplan using Lua?   Danny Nicholas
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Danny Nicholas
• [asterisk-users] create table in mysql using asterisk   Danny Nicholas
• [asterisk-users] 44Khz files in Asterisk 10   Danny Nicholas
• [asterisk-users] 44Khz files in Asterisk 10   Danny Nicholas
• [asterisk-users] Hang up phone after declined attended transfer   Danny Nicholas
• [asterisk-users] Linux Stun Server   Danny Nicholas
• [asterisk-users] Q: SIPNATtraversal.pdf   Danny Nicholas
• [asterisk-users] Speech recognition in asterisk using google voice API   Danny Nicholas
• [asterisk-users] SayDigits playback doesn't always work   Danny Nicholas
• [asterisk-users] SayDigits playback doesn't always work   Danny Nicholas
• [asterisk-users] How Can I configure the between call oneside IVR   Danny Nicholas
• [asterisk-users] How Can I configure the between call oneside IVR   Danny Nicholas
• [asterisk-users] /etc/init.d script and calling asterisk command line.   Danny Nicholas
• [asterisk-users] meetme with IVR   Danny Nicholas
• [asterisk-users] Pickup calls coming from queues   Danny Nicholas
• [asterisk-users] RFE idea for VM application   Danny Nicholas
• [asterisk-users] RFE idea for VM application   Danny Nicholas
• [asterisk-users] ChanSpy : how to know channel name ?   Danny Nicholas
• [asterisk-users] ChanSpy : how to know channel name ?   Danny Nicholas
• [asterisk-users] RFE idea for VM application   Danny Nicholas
• [asterisk-users] ChanSpy : how to know channel name ?   Danny Nicholas
• [asterisk-users] ChanSpy : how to know channel name ?   Danny Nicholas
• [asterisk-users] ChanSpy : how to know channel name ?   Danny Nicholas
• [asterisk-users] RFE idea for VM application   Danny Nicholas
• [asterisk-users] RFE idea for VM application   Danny Nicholas
• [asterisk-users] ChanSpy : how to know channel name ?   Danny Nicholas
• [asterisk-users] Is there a sip show equivelant.   Danny Nicholas
• [asterisk-users] ChanSpy : how to know channel name ?   Danny Nicholas
• [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality   Danny Nicholas
• [asterisk-users] sip reload and TCP transport.   Danny Nicholas
• [asterisk-users] upgraded 1.8.8.0 > 10.1.0-rc2: now db warnings   Danny Nicholas
• [asterisk-users] Cell Phone as a Queue member   Danny Nicholas
• [asterisk-users] Proposed changes to Asterisk release and support cycles   Danny Nicholas
• [asterisk-users] Proposed changes to Asterisk release and support cycles   Danny Nicholas
• [asterisk-users] Proposed changes to Asterisk release and support cycles   Danny Nicholas
• [asterisk-users] asterisk -> AGI (perl) -> sqlplus(oracle)   Dale Noll
• [asterisk-users] Real T1 trunk group...   Dale Noll
• [asterisk-users] Real T1 trunk group...   Dale Noll
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Dale Noll
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   John Novack
• [asterisk-users] SDP Issue   John Novack
• [asterisk-users] Issue with dahdi 2.5.0 and Digium HA8-B400M   Olivier
• [asterisk-users] Asterisk 1.8 - BRI D Channel going up and down every few seconds   Olivier
• [asterisk-users] Which QSIG variant and profiles does asterisk support ?   Olivier
• [asterisk-users] Noise in caller handset when dialing out (with dahdi 2.6.0)   Olivier
• [asterisk-users] Noise in caller handset when dialing out (with dahdi 2.6.0)   Olivier
• [asterisk-users] Noise in caller handset when dialing out (with dahdi 2.6.0) [SOLVED]   Olivier
• [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?   Olivier
• [asterisk-users] Anyone have a reliable T.38 Solution   Olivier
• [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?   Olivier
• [asterisk-users] OT - Which iceweasel plugin to play gsm sound files ?   Olivier
• [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?   Olivier
• [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED]   Olivier
• [asterisk-users] Anyone have a reliable T.38 Solution   Olivier
• [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?   Olivier
• [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED]   Olivier
• [asterisk-users] Anyone have a reliable T.38 Solution   Olivier
• [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ? [SOLVED]   Olivier
• [asterisk-users] Questions on hardware or software-based echo cancellation   Olivier
• [asterisk-users] Questions on hardware or software-based echo cancellation   Olivier
• [asterisk-users] How to check currently used libraries from command line ?   Olivier
• [asterisk-users] How to check currently used libraries from command line ?   Olivier
• [asterisk-users] Where to find meaning of /n in Local/6613 at from-queue/n ?   Olivier
• [asterisk-users] Where to find meaning of /n inLocal/6613 at from-queue/n ? [SOLVED]   Olivier
• [asterisk-users] Update callee num or name at caller display   Olivier
• [asterisk-users] OT - Configuring Freepbx's fax_process.pl to work with ssmtp   Olivier
• [asterisk-users] Update callee num or name at caller display   Olivier
• [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?   Olivier
• [asterisk-users] Efficient logging of PRI traffic for later analysis?   Olivier
• [asterisk-users] Pickup calls coming from queues   Olivier
• [asterisk-users] Pickup calls coming from queues   Olivier
• [asterisk-users] SendFax not sending AMI events   Olivier
• [asterisk-users] Dropping incompatible voice frame error   Kevin Oravits
• [asterisk-users] Is there any way to terminate async origination initialized by AMY?   Yaroslav Panych
• [asterisk-users] Asterisk 1.8.9.0 Now Available   Jason Parker
• [asterisk-users] Cordless SIP phone   Jason W. Parks
• [asterisk-users] RFC 5922 (TLS Certificates) and Asterisk   Daniel Pocock
• [asterisk-users] TLS problems - patch in Jira   Daniel Pocock
• [asterisk-users] CA Issued Certificates / TLS + SRTP   Daniel Pocock
• [asterisk-users] SRV record for non-standard SIP port?   Daniel Pocock
• [asterisk-users] which choice: asterisk-gui or freepbx?   Tom Poe
• [asterisk-users] best softphone for 2012?   Tom Poe
• [asterisk-users] best softphone for 2012?   Tom Poe
• [asterisk-users] dialplan -> dial command -> custom ringtone   Qqblog Qqblog
• [asterisk-users] 回覆︰ dialplan -> dial command -> custom ringtone   Qqblog Qqblog
• [asterisk-users] Edwin wants to partner with you!   Edwin Quijada
• [asterisk-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.   Zohair Raza
• [asterisk-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.   Zohair Raza
• [asterisk-users] DIALSTATUS Values   Zohair Raza
• [asterisk-users] how to set callerid in php AGI file.   Zohair Raza
• [asterisk-users] how to set callerid in php AGI file.   Zohair Raza
• [asterisk-users] how to set callerid in php AGI file.   Zohair Raza
• [asterisk-users] how to set callerid in php AGI file.   Zohair Raza
• [asterisk-users] Prepaid billing   Zohair Raza
• [asterisk-users] Prepaid billing   Zohair Raza
• [asterisk-users] Prepaid billing   Zohair Raza
• [asterisk-users] Change the caller's phone number   Phil Reynolds
• [asterisk-users] attended transfers going to wrong voicemail Asterisk 1.8 Polycom 650   Jeff Roberts
• [asterisk-users] Iax hold events in AMI 1.1   Alexandre Rodrigues
• [asterisk-users] dialplan -> dial command -> custom ringtone   Carlos Rojas
• [asterisk-users] asterisk problem sip   Carlos Rojas
• [asterisk-users] asterisk problem sip   Carlos Rojas
• [asterisk-users] How to make SIP guest call   Roland
• [asterisk-users] Problem connecting to 4569/UDP   Roland
• [asterisk-users] SayDigits playback doesn't always work   Roland
• [asterisk-users] SayDigits playback doesn't always work   Roland
• [asterisk-users] SayDigits playback doesn't always work   Roland
• [asterisk-users] Ringing agents cell as an alert?   Todd Routhier
• [asterisk-users] Ringing agents cell as an alert?   Todd Routhier
• [asterisk-users] Answering call from queue, then put back in queue?   Todd Routhier
• [asterisk-users] Answering call from queue, then put back in queue?   Todd Routhier
• [asterisk-users] ISDN E1, after electrical disconnected is not becoming UP, IRQ misses: 1   Shaun Ruffell
• [asterisk-users] Noise in caller handset when dialing out (with dahdi 2.6.0)   Shaun Ruffell
• [asterisk-users] Noise in caller handset when dialing out (with dahdi 2.6.0)   Shaun Ruffell
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Shaun Ruffell
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Shaun Ruffell
• [asterisk-users] FAX Installation in Asterisk   Ruben Rögels
• [asterisk-users] FAX Installation in Asterisk   Ruben Rögels
• [asterisk-users] FAX Installation in Asterisk   Ruben Rögels
• [asterisk-users] How to check currently used libraries from command line ?   Ruben Rögels
• [asterisk-users] asterisk not connecting to sipgate / NAT related issue?   Ruben Rögels
• [asterisk-users] Anyone have a reliable T.38 Solution   Frank Sautter
• [asterisk-users] Update callee num or name at caller display   Gunnar Schaller
• [asterisk-users] Update callee num or name at caller display   Gunnar Schaller
• [asterisk-users] Update callee num or name at caller display   Gunnar Schaller
• [asterisk-users] Update callee num or name at caller display   Gunnar Schaller
• [asterisk-users] Voicemail weirdness after upgrade   Paul Schenkeveld
• [asterisk-users] Asterisk rewrites "From" header when CALLERID(num-pres)=prohib_passed_screen is set   Stefan Schmidt
• [asterisk-users] Ringing agents cell as an alert?   James Sharp
• [asterisk-users] Ringing agents cell as an alert?   James Sharp
• [asterisk-users] Mark queue agent as away   James Sharp
• [asterisk-users] create table in mysql using asterisk   James Sharp
• [asterisk-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.   Kaushal Shriyan
• [asterisk-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.   Kaushal Shriyan
• [asterisk-users] dialplan problem : not including context   Andreas Sikkema
• [asterisk-users] Sporadic one way audio problem   Andreas Sikkema
• [asterisk-users] Cordless SIP phone   Andreas Sikkema
• [asterisk-users] MFCR2 Long distance calls not connected   Moises Silva
• [asterisk-users] asterisk 1.8 codec negotiation   José Pablo Méndez Soto
• [asterisk-users] Where are the fax instructions?   José Pablo Méndez Soto
• [asterisk-users] Where are the fax instructions?   José Pablo Méndez Soto
• [asterisk-users] Best non polycom SIP conference room phone   José Pablo Méndez Soto
• [asterisk-users] Change port from 5060 on Snom phone   José Pablo Méndez Soto
• [asterisk-users] Why write your dialplan using Lua?   José Pablo Méndez Soto
• [asterisk-users] Where are the fax instructions?   José Pablo Méndez Soto
• [asterisk-users] Where are the fax instructions?   José Pablo Méndez Soto
• [asterisk-users] Why write your dialplan using Lua?   José Pablo Méndez Soto
• [asterisk-users] Why write your dialplan using Lua?   José Pablo Méndez Soto
• [asterisk-users] Change port from 5060 on Snom phone   José Pablo Méndez Soto
• [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?   José Pablo Méndez Soto
• [asterisk-users] Problems faced in load testing of asterisk   José Pablo Méndez Soto
• [asterisk-users] Congestion outbound only with ATA boxes   Royce Souther
• [asterisk-users] Streaming Music to 75 callers ..   Sriram
• [asterisk-users] Asterisk rewrites "From" header when CALLERID(num-pres)=prohib_passed_screen is set   Paris Stamatopoulos
• [asterisk-users] Peer doesn't answer   Arthur Stanfield
• [asterisk-users] [NAT] SSH vs. OpenVPN?   Arthur Stanfield
• [asterisk-users] Best non polycom SIP conference room phone   Jamie A. Stapleton
• [asterisk-users] Best non polycom SIP conference room phone   Jamie A. Stapleton
• [asterisk-users] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?   Johann Steinwendtner
• [asterisk-users] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?   Johann Steinwendtner
• [asterisk-users] Asterisk won't start - trap invalid opcode   A J Stiles
• [asterisk-users] Asterisk won't start - trap invalid opcode   A J Stiles
• [asterisk-users] Connecting to an Old Phone System   A J Stiles
• [asterisk-users] create table in mysql using asterisk   A J Stiles
• [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED]   A J Stiles
• [asterisk-users] How to check currently used libraries from command line ?   A J Stiles
• [asterisk-users] Starting things off without a dial tone   A J Stiles
• [asterisk-users] /etc/init.d script and calling asterisk command line.   A J Stiles
• [asterisk-users] Efficient logging of PRI traffic for later analysis?   A J Stiles
• [asterisk-users] Cordless SIP phone   A J Stiles
• [asterisk-users] Cordless SIP phone   A J Stiles
• [asterisk-users] RFE idea for VM application   A J Stiles
• [asterisk-users] ConfBridge no audio problem   Roi Stork
• [asterisk-users] no audio using g729A for Cisco AS5300 sip peer   Roi Stork
• [asterisk-users] Cisco AS5300 and Digium g729A codec   Roi Stork
• [asterisk-users] Cisco AS5300 and Digium g729A codec   Roi Stork
• [asterisk-users] Cisco AS5300 and Digium g729A codec   Roi Stork
• [asterisk-users] Cisco AS5300 and Digium g729A codec   Roi Stork
• [asterisk-users] local channels and g729a voice quality   Roi Stork
• [asterisk-users] local channels and g729a voice quality   Roi Stork
• [asterisk-users] dialplan problem : not including context   Administrator TOOTAI
• [asterisk-users] atx timeout - play xferfailsound   John Taylor
• [asterisk-users] vigor 2920 problems   John Taylor
• [asterisk-users] Avaya 4610sw IP Phone   Jonn Taylor
• [asterisk-users] DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0 Released   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.8.2 and 10.0.1 Now Available (Security Release)   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.9.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 10.1.0 Now Available   Asterisk Development Team
• [asterisk-users] AST-2012-001: SRTP Video Remote Crash Vulnerability   Asterisk Security Team
• [asterisk-users] Asterisk 1.4 and configuration to be via Database instead of conf files   Brynjolfur Thorvardsson
• [asterisk-users] Asterisk won't start - trap invalid opcode   Duncan Turnbull
• [asterisk-users] Asterisk won't start - trap invalid opcode   Duncan Turnbull
• [asterisk-users] Asterisk won't start - trap invalid opcode   Duncan Turnbull
• [asterisk-users] Asterisk won't start - trap invalid opcode   Duncan Turnbull
• [asterisk-users] Asterisk won't start - trap invalid opcode   Duncan Turnbull
• [asterisk-users] Anyone have a reliable T.38 Solution   Steve Underwood
• [asterisk-users] Anyone have a reliable T.38 Solution   Steve Underwood
• [asterisk-users] Anyone have a reliable T.38 Solution   Steve Underwood
• [asterisk-users] FAX Installation in Asterisk   Steve Underwood
• [asterisk-users] local channels and g729a voice quality   Steve Underwood
• [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?   Steve Underwood
• [asterisk-users] SIP hardphone with dual gigabit ethernet ports   Vieri
• [asterisk-users] User hit f to disconnect call.   Vieri
• [asterisk-users] User hit f to disconnect call.   Vieri
• [asterisk-users] video mail is not store   Alex Vishnev
• [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?   Kristijan Vrban
• [asterisk-users] Anyone have a reliable T.38 Solution   Ryan Wagoner
• [asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller   Ryan Wagoner
• [asterisk-users] Best non polycom SIP conference room phone   Ryan Wagoner
• [asterisk-users] Hang up phone after declined attended transfer   Ryan Wagoner
• [asterisk-users] Hang up phone after declined attended transfer   Ryan Wagoner
• [asterisk-users] Too many open files   Chad Wallace
• [asterisk-users] Analoge and E1 ports   Karsten Wemheuer
• [asterisk-users] Question on system command 1.4.43   Eric Wieling
• [asterisk-users] Set Call Codec in extension.conf   Eric Wieling
• [asterisk-users] Set Call Codec in extension.conf   Eric Wieling
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Eric Wieling
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Eric Wieling
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Eric Wieling
• [asterisk-users] asterisk 1.8.8 - caller ID not working.   Eric Wieling
• [asterisk-users] Asterisk 10.0 & 1.4 - iax codec are not compatible   Eric Wieling
• [asterisk-users] [SOLVED] Asterisk 10.0 & 1.4 - iax codec are not compatible   Eric Wieling
• [asterisk-users] [SOLVED] Asterisk 10.0 & 1.4 - iax codec are not compatible   Eric Wieling
• [asterisk-users] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?   Eric Wieling
• [asterisk-users] SayDigits playback doesn't always work   Eric Wieling
• [asterisk-users] Update callee num or name at caller display   Eric Wieling
• [asterisk-users] Problem answering phone   Eric Wieling
• [asterisk-users] asterisk does not detect menus   Eric Wieling
• [asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?   Johan Wilfer
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Johan Wilfer
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Johan Wilfer
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Johan Wilfer
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Johan Wilfer
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Johan Wilfer
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Johan Wilfer
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Johan Wilfer
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Johan Wilfer
• [asterisk-users] Dahdi for meetme on AMD64 arch?   Johan Wilfer
• [asterisk-users] meetme - Unable to write frame to channel   Johan Wilfer
• [asterisk-users] play sound file   Johan Wilfer
• [asterisk-users] upgraded 1.8.8.0 > 10.1.0-rc2: now db warnings   Terry Wilson
• [asterisk-users] best softphone for 2012?   Hans Witvliet
• [asterisk-users] CDR into ical?   Jay R. Worthington
• [asterisk-users] Speech recognition in asterisk using google voice API   Lefteris Zafiris
• [asterisk-users] Speech recognition in asterisk using google voice API   Lefteris Zafiris
• [asterisk-users] Speech recognition in asterisk using google voice API   Lefteris Zafiris
• [asterisk-users] Speech recognition in asterisk using google voice API   Lefteris Zafiris
• [asterisk-users] Speech recognition in asterisk using google voice API   Lefteris Zafiris
• [asterisk-users] Speech recognition in asterisk using google voice API   Lefteris Zafiris
• [asterisk-users] Speech recognition in asterisk using google voice API   Lefteris Zafiris
• [asterisk-users] Speech recognition in asterisk using google voice API   Lefteris Zafiris
• [asterisk-users] Speech recognition in asterisk using google voice API   Lefteris Zafiris
• [asterisk-users] Speech recognition in asterisk using google voice API   Lefteris Zafiris
• [asterisk-users] Speech recognition in asterisk using google voice API   Lefteris Zafiris
• [asterisk-users] 44Khz files in Asterisk 10   Lefteris Zafiris
• [asterisk-users] 44Khz files in Asterisk 10   Lefteris Zafiris
• [asterisk-users] Speech recognition in asterisk using google voice API   Lefteris Zafiris
• [asterisk-users] Best non polycom SIP conference room phone   Bryant Zimmerman
• [asterisk-users] Linux Stun Server   Bryant Zimmerman
• [asterisk-users] Best non polycom SIP conference room phone   Bryant Zimmerman
• [asterisk-users] SIP and NAT best practices since recent changes?   Bryant Zimmerman
• [asterisk-users] Macro vs sub   Bryant Zimmerman
• [asterisk-users] /etc/init.d script and calling asterisk command line.   Bryant Zimmerman
• [asterisk-users] /etc/init.d script and calling asterisk command line.   Bryant Zimmerman
• [asterisk-users] asterisk does not detect menus   Bryant Zimmerman
• [asterisk-users] Is there a sip show equivelant.   Bryant Zimmerman
• [asterisk-users] sip reload and TCP transport.   Bryant Zimmerman
• [asterisk-users] TCP transport and BLF   Bryant Zimmerman
• [asterisk-users] fall back to inband DTMF?   Bryant Zimmerman
• [asterisk-users] fall back to inband DTMF?   Bryant Zimmerman
• [asterisk-users] Proposed changes to Asterisk release and support cycles   Bryant Zimmerman
• [asterisk-users] Proposed changes to Asterisk release and support cycles   Bryant Zimmerman
• [asterisk-users] unsubscribe   Dietmar Zlabinger
• [asterisk-users] pickup group   Sinkovicz Zoltan
• [asterisk-users] Asterisk as UAC: How to put call OnHold   Johannes Zweng
• [asterisk-users] Asterisk as UAC: How to put call OnHold   Johannes Zweng
• [asterisk-users] Asterisk as UAC: How to put call OnHold   Johannes Zweng
• [asterisk-users] Asterisk as UAC: How to put call OnHold   Johannes Zweng
• [asterisk-users] Voicemail weirdness after upgrade   --[ UxBoD ]--
• [asterisk-users] SDP Issue   --[ UxBoD ]--
• [asterisk-users] SDP Issue   --[ UxBoD ]--
• [asterisk-users] SDP Issue   --[ UxBoD ]--
• [asterisk-users] play sound file   amit anand
• [asterisk-users] Queue option 'R'   bakko
• [asterisk-users] Where to find meaning of /n inLocal/6613 at from-queue/n ?   bakko
• [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality   bakko
• [asterisk-users] [NAT] SSH vs. OpenVPN?   bakko
• [asterisk-users] performance/memory   virendra bhati
• [asterisk-users] Set Call type in dial plan   virendra bhati
• [asterisk-users] Set Call type in dial plan   virendra bhati
• [asterisk-users] Set Call type in dial plan   virendra bhati
• [asterisk-users] Failed to authenticate on INVITE to Anonymous   virendra bhati
• [asterisk-users] Failed to authenticate on INVITE to Anonymous   virendra bhati
• [asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??   virendra bhati
• [asterisk-users] how to set callerid in php AGI file.   virendra bhati
• [asterisk-users] how to set callerid in php AGI file.   virendra bhati
• [asterisk-users] Prepaid billing   virendra bhati
• [asterisk-users] Prepaid billing   virendra bhati
• [asterisk-users] Failed to Allocate port for RTP instance   virendra bhati
• [asterisk-users] Asterisk to support Dialogic Cards   virendra bhati
• [asterisk-users] Asterisk to support Dialogic Cards   virendra bhati
• [asterisk-users] View # active calls in a context   virendra bhati
• [asterisk-users] Force CDR to be written.   virendra bhati
• [asterisk-users] Strange how Asterisk know the updated information of log   virendra bhati
• [asterisk-users] Strange how Asterisk know the updated information of log   virendra bhati
• [asterisk-users] Strange how Asterisk know the updated information of log   virendra bhati
• [asterisk-users] asterisk 1.8 codec negotiation   covici at ccs.covici.com
• [asterisk-users] asterisk 1.8 codec negotiation   covici at ccs.covici.com
• [asterisk-users] tcp version of toronto - osaka doesn't work   sean darcy
• [asterisk-users] tcp version of toronto - osaka doesn't work   sean darcy
• [asterisk-users] tcp version of toronto - osaka doesn't work   sean darcy
• [asterisk-users] tcp version of toronto - osaka doesn't work   sean darcy
• [asterisk-users] Speech recognition in asterisk using google voice API   sean darcy
• [asterisk-users] Failed to authenticate on INVITE to Anonymous   sean darcy
• [asterisk-users] best softphone for 2012?   sean darcy
• [asterisk-users] 10.1.0-rc1 : WARNING: abstract_jb.c:384 jb_get_and_deliver: AST_JB_IMPL_NOFRAME   sean darcy
• [asterisk-users] upgraded 1.8.8.0 > 10.1.0-rc2: now db warnings   sean darcy
• [asterisk-users] upgraded 1.8.8.0 > 10.1.0-rc2: now db warnings   sean darcy
• [asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??   shalu dhamija
• [asterisk-users] Problems faced in load testing of asterisk   shalu dhamija
• [asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??   shalu dhamija
• [asterisk-users] odbc storage for video message   shalu dhamija
• [asterisk-users] Failed to Allocate port for RTP instance   shalu dhamija
• [asterisk-users] Failed to Allocate port for RTP instance   shalu dhamija
• [asterisk-users] Sip Registration Hijacking   eherr
• [asterisk-users] Sip Registration Hijacking   eherr
• [asterisk-users] Cordless SIP phone   eherr
• [asterisk-users] Cordless SIP phone   eherr
• [asterisk-users] Cordless SIP phone   eherr
• [asterisk-users] Sip Registration Hijacking   eherr
• [asterisk-users] Sip Registration Hijacking   eherr
• [asterisk-users] Sip Registration Hijacking   eherr
• [asterisk-users] Sip Registration Hijacking   eherr
• [asterisk-users] fail2ban restarts   eherr
• [asterisk-users] Codec   Dustin fails
• [asterisk-users] GoAutoDialer, ViciDial and Vicidial group   bilal ghayyad
• [asterisk-users] ISDN E1, after electrical disconnected is not becoming UP, IRQ misses: 1   bilal ghayyad
• [asterisk-users] Analoge and E1 ports   bilal ghayyad
• [asterisk-users] Asterisk 1.4 and configuration to be via Database instead of conf files   bilal ghayyad
• [asterisk-users] Asterisk 1.4 and configuration to be via Database instead of conf files   bilal ghayyad
• [asterisk-users] Speech recognition in asterisk using google voice API   isrlgb at gmail.com
• [asterisk-users] SIP hardphone with dual gigabit ethernet ports   isrlgb at gmail.com
• [asterisk-users] Rami   gokulnath
• [asterisk-users] Client - registers but unreachable   white hat
• [asterisk-users] Client - registers but unreachable   white hat
• [asterisk-users] t38modem v2, which version or patch of asterisk?   Cyber.fox1 at infinito.it
• [asterisk-users] registration not authorized - stale nonce   asterisk jobs
• [asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?   asterisk jobs
• [asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?   asterisk jobs
• [asterisk-users] Weird IPs in Fail2ban list   asterisk jobs
• [asterisk-users] FAX Installation in Asterisk   mahesh katta
• [asterisk-users] FAX Installation in Asterisk   mahesh katta
• [asterisk-users] FAX Installation in Asterisk   mahesh katta
• [asterisk-users] FAX Installation in Asterisk   mahesh katta
• [asterisk-users] FAX Installation in Asterisk   mahesh katta
• [asterisk-users] FAX Installation in Asterisk   mahesh katta
• [asterisk-users] FAX Installation in Asterisk   mahesh katta
• [asterisk-users] How Can I configure the between call oneside IVR   mahesh katta
• [asterisk-users] How Can I configure the between call oneside IVR   mahesh katta
• [asterisk-users] How Can I configure the between call oneside IVR   mahesh katta
• [asterisk-users] meetme with IVR   mahesh katta
• [asterisk-users] meetme with IVR   mahesh katta
• [asterisk-users] meetme with IVR   mahesh katta
• [asterisk-users] Problem connecting to 4569/UDP   kazabe
• [asterisk-users] Problem connecting to 4569/UDP   kazabe
• [asterisk-users] echo & audio delay in SIP VOIP   mahendra
• [asterisk-users] 1.6 and 1.8   mattias
• [asterisk-users] From address missing 'sip:', using it anyway   motty.cruz
• [asterisk-users] asterisk does not detect menus   motty.cruz
• [asterisk-users] asterisk does not detect menus   motty.cruz
• [asterisk-users] asterisk does not detect menus   motty.cruz
• [asterisk-users] Asterisk rewrites "From" header when CALLERID(num-pres)=prohib_passed_screen is set   effie mouzeli
• [asterisk-users] Asterisk rewrites "From" header when CALLERID(num-pres)=prohib_passed_screen is set   effie mouzeli
• [asterisk-users] Queue option 'R'   georg at riseup.net
• [asterisk-users] Sporadic one way audio problem   georg at riseup.net
• [asterisk-users] Queue option 'R'   georg at riseup.net
• [asterisk-users] Sporadic one way audio problem   georg at riseup.net
• [asterisk-users] Layer2 Down in BRI connection   khalid touati
• [asterisk-users] Layer2 Down in BRI connection   khalid touati
• [asterisk-users] question sangoma vs digium   James zhu
• [asterisk-users] Best non polycom SIP conference room phone   BryantZ at zktech.com

Last message date: Tue Jan 31 19:12:06 CST 2012
Archived on: Tue Jan 31 19:18:05 CST 2012

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