January 2012 Archives by thread
Starting: Sun Jan 1 06:22:50 CST 2012
Ending: Tue Jan 31 19:12:06 CST 2012
Messages: 902
- [asterisk-users] 481 Call leg/transaction does not exists Status Response
Elliot Murdock
- [asterisk-users] 1.6 and 1.8
Elliot Murdock
- [asterisk-users] GoAutoDialer, ViciDial and Vicidial group
bilal ghayyad
- [asterisk-users] asterisk 1.8 codec negotiation
covici at ccs.covici.com
- [asterisk-users] Client - registers but unreachable
white hat
- [asterisk-users] tcp version of toronto - osaka doesn't work
sean darcy
- [asterisk-users] Help_video voice mail not retriev properly
Durgesh Mishra
- [asterisk-users] Problem w/ PC port on Polycom 335
Mike Diehl
- [asterisk-users] dialplan -> dial command -> custom ringtone
Qqblog Qqblog
- [asterisk-users] Asterisk 1.8 - BRI D Channel going up and down every few seconds
Marco Mooijekind
- [asterisk-users] Issue with dahdi 2.5.0 and Digium HA8-B400M
Olivier
- [asterisk-users] Using Asterisk as a softphone
Christian Jaeger
- [asterisk-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Kaushal Shriyan
- [asterisk-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Zohair Raza
- [asterisk-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Kaushal Shriyan
- [asterisk-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Zohair Raza
- [asterisk-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Patrick Lists
- [asterisk-users] Question on system command 1.4.43
Jerry Geis
- [asterisk-users] NAT/IPTABLES workarounds
Danny Nicholas
- [asterisk-users] Problem connecting to 4569/UDP
kazabe
- [asterisk-users] ISDN E1, after electrical disconnected is not becoming UP, IRQ misses: 1
bilal ghayyad
- [asterisk-users] Speech recognition in asterisk using google voice API
Lefteris Zafiris
- [asterisk-users] Which QSIG variant and profiles does asterisk support ?
Olivier
- [asterisk-users] From address missing 'sip:', using it anyway
motty.cruz
- [asterisk-users] question sangoma vs digium
Agustina Berretta
- [asterisk-users] DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0 Released
Asterisk Development Team
- [asterisk-users] Server-to-server BLF
Ronald Cepres
- [asterisk-users] Where are the fax instructions?
José Pablo Méndez Soto
- [asterisk-users] Video trancoding not done.
Durgesh Mishra
- [asterisk-users] Asterisk1.8 support video trancoding ?
Durgesh Mishra
- [asterisk-users] Best non polycom SIP conference room phone
Bryant Zimmerman
- [asterisk-users] question on CDR
Jerry Geis
- [asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller
Douglas Mortensen
- [asterisk-users] STOP loading extensions.ael
Joseph
- [asterisk-users] asterisk 1.8.8 - caller ID not working.
Joseph
- [asterisk-users] which choice: asterisk-gui or freepbx?
Tom Poe
- [asterisk-users] calling specific 1800-number not going through.
Joseph
- [asterisk-users] Connecting to an Old Phone System
Dan Journo
- [asterisk-users] Change port from 5060 on Snom phone
Ishfaq Malik
- [asterisk-users] Why write your dialplan using Lua?
José Pablo Méndez Soto
- [asterisk-users] best softphone for 2012?
Tom Poe
- [asterisk-users] Asterisk 10.0 & 1.4 - iax codec are not compatible
Joseph
- [asterisk-users] Couple of questions: SIP ALG, allowguest=no
Gilles
- [asterisk-users] cached VMI on manual voicemail update
Tzafrir Cohen
- [asterisk-users] video mail is not store
Durgesh Mishra
- [asterisk-users] create table in mysql using asterisk
Eyal
- [asterisk-users] Asterisk as register server through OpenSIPS
Ronald Cepres
- [asterisk-users] Cisco AS5300 and Digium g729A codec
Roi Stork
- [asterisk-users] DEBUG Message
Elliot Murdock
- [asterisk-users] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?
Alex Villacís Lasso
- [asterisk-users] Noise in caller handset when dialing out (with dahdi 2.6.0)
Olivier
- [asterisk-users] 44Khz files in Asterisk 10
Danny Nicholas
- [asterisk-users] Noise in caller handset when dialing out (with dahdi 2.6.0) [SOLVED]
Olivier
- [asterisk-users] Linux Stun Server
Bryant Zimmerman
- [asterisk-users] Best non polycom SIP conference room phone
Bryant Zimmerman
- [asterisk-users] Odd DTMF problem when receiving calls
Christopher David Howie
- [asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??
shalu dhamija
- [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
Olivier
- [asterisk-users] Problems faced in load testing of asterisk
shalu dhamija
- [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
Olivier
- [asterisk-users] Q: SIPNATtraversal.pdf
Matthias Apitz
- [asterisk-users] Iax hold events in AMI 1.1
Alexandre Rodrigues
- [asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??
shalu dhamija
- [asterisk-users] SIP and NAT best practices since recent changes?
Bryant Zimmerman
- [asterisk-users] Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1
Alex Villacís Lasso
- [asterisk-users] Problems with codec translation when using Monitor and MixMonitor
Daniel - Asterisk
- [asterisk-users] how to set callerid in php AGI file.
virendra bhati
- [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ? [SOLVED]
Olivier
- [asterisk-users] Call abandoned from queue not showing in CDR (possible bug)
Ishfaq Malik
- [asterisk-users] t38modem v2, which version or patch of asterisk?
Cyber.fox1 at infinito.it
- [asterisk-users] odbc storage for video message
shalu dhamija
- [asterisk-users] dialplan problem : not including context
Jonas Kellens
- [asterisk-users] SIP hardphone with dual gigabit ethernet ports
Vieri
- [asterisk-users] Stuck DAHDI Channels
Antonio Modesto
- [asterisk-users] Queue option 'R'
georg at riseup.net
- [asterisk-users] Sporadic one way audio problem
georg at riseup.net
- [asterisk-users] Asterisk as UAC: How to put call OnHold
Johannes Zweng
- [asterisk-users] CDR into ical?
Jay R. Worthington
- [asterisk-users] AstLinux 1.01 Released
Darrick Hartman
- [asterisk-users] Peer doesn't answer
Arlen Nascimento
- [asterisk-users] local channels and g729a voice quality
Roi Stork
- [asterisk-users] How to check currently used libraries from command line ?
Olivier
- [asterisk-users] Real T1 trunk group...
Louis Carreiro
- [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
Gunnar Schaller
- [asterisk-users] Starting things off without a dial tone
A J Stiles
- [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] OT - Configuring Freepbx's fax_process.pl to work with ssmtp
Olivier
- [asterisk-users] Core file created in /tmp
Jonas Kellens
- [asterisk-users] Macro vs sub
Jonas Kellens
- [asterisk-users] Macro vs sub
Bryant Zimmerman
- [asterisk-users] /etc/init.d script and calling asterisk command line.
Bryant Zimmerman
- [asterisk-users] Prepaid billing
Zohair Raza
- [asterisk-users] Problem answering phone
Mike Diehl
- [asterisk-users] /etc/init.d script and calling asterisk command line.
Bryant Zimmerman
- [asterisk-users] SIP trunk call initiated as Anonymous at anonymous.invalid
Gordon Messmer
- [asterisk-users] Failed to Allocate port for RTP instance
shalu dhamija
- [asterisk-users] Dahdi for meetme on AMD64 arch?
Johan Wilfer
- [asterisk-users] Installing the 3.1 sources of Kernel with Asterisk
Christian
- [asterisk-users] Failed to Allocate port for RTP instance
shalu dhamija
- [asterisk-users] Does Asterisk permit multiple registrations to the same host?
Frank Church
- [asterisk-users] Huh? Local is being asked to answer?
Agustina Berretta
- [asterisk-users] Voicemail weirdness after upgrade
Paul Schenkeveld
- [asterisk-users] Asterisk 1.8 - SIP losing registration
Joseph
- [asterisk-users] Efficient logging of PRI traffic for later analysis?
Tony Mountifield
- [asterisk-users] Sip Registration Hijacking
eherr
- [asterisk-users] Asterisk NOT in the media path
Jonas Kellens
- [asterisk-users] Pickup calls coming from queues
Niccolò Belli
- [asterisk-users] 10.1.0-rc1 : WARNING: abstract_jb.c:384 jb_get_and_deliver: AST_JB_IMPL_NOFRAME
sean darcy
- [asterisk-users] View # active calls in a context
Michelle Dupuis
- [asterisk-users] Force CDR to be written.
Jim DeVito
- [asterisk-users] Analoge and E1 ports
bilal ghayyad
- [asterisk-users] Chan_Mobile Nokia E51, csr bt dogle, Voice OK but no SMS Support ?
Din Assegaf
- [asterisk-users] SIP - connected line has changed. Saving it until answer for IAX2/iaxy
Joseph
- [asterisk-users] Avaya 4610sw IP Phone
Aamir Chougule
- [asterisk-users] Cordless SIP phone
eherr
- [asterisk-users] SDP Issue
--[ UxBoD ]--
- [asterisk-users] Timing Slips CRC & E-Bit Errors - Asterisk - Trixbox 2.8.0.4
Liban Abdi
- [asterisk-users] asterisk does not detect menus
motty.cruz
- [asterisk-users] asterisk does not detect menus
Bryant Zimmerman
- [asterisk-users] ConfBridge details
Jeremy Kister
- [asterisk-users] ChanSpy : how to know channel name ?
Jonas Kellens
- [asterisk-users] RFE idea for VM application
Phil Daws
- [asterisk-users] allowguest = yes? no?
Gilles
- [asterisk-users] Is there a sip show equivelant.
Bryant Zimmerman
- [asterisk-users] Signalling and Media Configuration
Ryan Icasiano
- [asterisk-users] Blocking in: ast_waitfor_nandfds
Ishfaq Malik
- [asterisk-users] play sound file
Eyal
- [asterisk-users] User hit f to disconnect call.
Vieri
- [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality
Tim Nelson
- [asterisk-users] Too many open files
Mike Diehl
- [asterisk-users] unsubscribe
Dietmar Zlabinger
- [asterisk-users] upgraded 1.8.8.0 > 10.1.0-rc2: now db warnings
sean darcy
- [asterisk-users] sip reload and TCP transport.
Bryant Zimmerman
- [asterisk-users] Asterisk 1.4 and configuration to be via Database instead of conf files
bilal ghayyad
- [asterisk-users] TCP transport and BLF
Bryant Zimmerman
- [asterisk-users] process_sdp: Unsupported SDP media type in offer: audio , Failing due to no acceptable offer found
Din Assegaf
- [asterisk-users] SendFax not sending AMI events
Mike Diehl
- [asterisk-users] fail2ban restarts
eherr
- [asterisk-users] atx timeout - play xferfailsound
John Taylor
- [asterisk-users] Asterisk 1.4 and configuration to be via Database instead of conf files
bilal ghayyad
- [asterisk-users] vigor 2920 problems
John Taylor
- [asterisk-users] CA Issued Certificates / TLS + SRTP
Stuart Elvish
- [asterisk-users] fall back to inband DTMF?
Bryant Zimmerman
- [asterisk-users] RFC 5922 (TLS Certificates) and Asterisk
Daniel Pocock
- [asterisk-users] TLS problems - patch in Jira
Daniel Pocock
- [asterisk-users] fall back to inband DTMF?
Bryant Zimmerman
- [asterisk-users] AMI - Getting Event of QueueAgents WrapupTime State
Karsten Asche
- [asterisk-users] Cell Phone as a Queue member
Niccolò Belli
- [asterisk-users] [NAT] SSH vs. OpenVPN?
Gilles
- [asterisk-users] SRV record for non-standard SIP port?
Gilles
- [asterisk-users] Proposed changes to Asterisk release and support cycles
Kevin P. Fleming
- [asterisk-users] Deadlock detected in asterisk-1.8.9.0 x86_64
Alex Villacís Lasso
- [asterisk-users] Proposed changes to Asterisk release and support cycles
Bryant Zimmerman
- [asterisk-users] Experience with Eicon Diva PRO 3.0?
Michelle Konzack
- [asterisk-users] Proposed changes to Asterisk release and support cycles
Bryant Zimmerman
- [asterisk-users] Congestion outbound only with ATA boxes
Royce Souther
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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