November 2011 Archives by thread
Starting: Tue Nov 1 01:58:22 CDT 2011
Ending: Wed Nov 30 22:46:14 CST 2011
Messages: 741
- [asterisk-users] custom automated meeting
Thanasis
- [asterisk-users] Nat Phone in Asterisk 10
Anton Kvashenkin
- [asterisk-users] bug in queuemanager?
Henry Dogger
- [asterisk-users] Problem with Atxfer for the calling party
Antonio Modesto
- [asterisk-users] Issues with Xorcom astribanks when running Ubuntu 11.10 (oneiric) as a quemu/libvirt guest
Eric van der Vlist
- [asterisk-users] custom automated meeting
Yaroslav Panych
- [asterisk-users] State of Asterisk+Virtualization+Timing
Tim Nelson
- [asterisk-users] FFA - Asterisk 1.6.2.6
Christian Tardif
- [asterisk-users] Option 'd' of application Dial not working in 1.8.8-rc2
Karsten Wemheuer
- [asterisk-users] libpri : Q.931 Called Party Number interpreted as empty
Philippe Sultan
- [asterisk-users] Unable to build sip pvt data - Switching equipment congestion
Jonas Kellens
- [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks
Olivier
- [asterisk-users] duration limits in Dial() not being enforced at correct time
Bryant Zimmerman
- [asterisk-users] Asterisk as SoftSwitch - Hardware
Sunny
- [asterisk-users] 2 pbxes
mattias
- [asterisk-users] DID from Direct from Telco
Nick Khamis
- [asterisk-users] [SOLVED] custom automated meeting
Thanasis
- [asterisk-users] DID from Direct from Telco
Bryant Zimmerman
- [asterisk-users] problem when exiting from "record file" function without pressing the escape digit
Yaprak Ayazoglu
- [asterisk-users] 9. any live queue monitor recommendation? (Jean Chassoul) chassoul at gmail.com
Anthony Laudini
- [asterisk-users] DID from Direct from Telco
Bryant Zimmerman
- [asterisk-users] DID from Direct from Telco
Bryant Zimmerman
- [asterisk-users] Reporting for Asterisk Call Center
bilal ghayyad
- [asterisk-users] Where do I find error message descriptions?
Thorben Jensen
- [asterisk-users] Dahdi complete 2.5.0.1 - dahdi_dummy not compiled
Administrator TOOTAI
- [asterisk-users] 1.8.8.0-rc2 Missing core functions.
Bryant Zimmerman
- [asterisk-users] 4 sec delay in voice menu (asterisk)
Albert
- [asterisk-users] Many SIP-480 responses
Tobias Steen
- [asterisk-users] Asterisk 1.6.2.20 lost registration bug with NAT keep-alive
Carlos Alvarez
- [asterisk-users] DTMF issue with 1.8.6.0 and SIP Trunks
JR Richardson
- [asterisk-users] IAX2 availability testing
Jaap Winius
- [asterisk-users] Asterisk 10.0.0-rc1 Now Available
Asterisk Development Team
- [asterisk-users] Frequent Asterisk Restarts
Mike Diehl
- [asterisk-users] FXS - Power Alarms
Tim Nelson
- [asterisk-users] What the variable that return the IP Phone username to use it for AddQueueMember
bilal ghayyad
- [asterisk-users] 1.8.7.0 crashing : Can't send 10 type frames with SIP write
sean darcy
- [asterisk-users] Text to speech modules (espeak, flite)
Lefteris Zafiris
- [asterisk-users] 10.0.0-rc1: won't start: "empty buf size"
sean darcy
- [asterisk-users] 10.0.0-rc1: dahdi doesn't see card
sean darcy
- [asterisk-users] Music on Hold does not " Kick-in" until second try , on outgoing calls.
Guy Gold
- [asterisk-users] shared_lastcall for 1.4.42
Jason Marble
- [asterisk-users] Call to Asterisk registered sofphone from an independent unregistered Endpoint
Amar Akshat
- [asterisk-users] unavailable state not reported to Cisco SPA50X phone
Linux
- [asterisk-users] TE122
Jerry Geis
- [asterisk-users] Becoming a CLEC
eherr
- [asterisk-users] Becoming a CLEC
Bryant Zimmerman
- [asterisk-users] How do extensions "stay registered"
Douglas Mortensen
- [asterisk-users] trouble with sip connection and registration
sean darcy
- [asterisk-users] Goto Queue, does not work, it should play message or any thing
bilal ghayyad
- [asterisk-users] Forcing a CODEC
Jaap Winius
- [asterisk-users] Standard UIDs, especially for asterisk?
Tony Mountifield
- [asterisk-users] polycom soundpint ip650 question
eherr
- [asterisk-users] Polycom Attended Transfer
eherr
- [asterisk-users] Grandstream HT503 colgado
bakko
- [asterisk-users] Use Polycom FX with Asterisk
Malvin Rito
- [asterisk-users] AMI: anything to glue originate to events?
giovanni.v
- [asterisk-users] Fax not detected by Asterisk
ik
- [asterisk-users] Cutting noise and voice
Jayson Rowe
- [asterisk-users] 10-rc2: how to debug dropped calls?
sean darcy
- [asterisk-users] DTMF dropping in Read Command
Danny Nicholas
- [asterisk-users] Question about Read() application
Danny Nicholas
- [asterisk-users] 2 same sip extension number on 2 asterisk - call not passing on certain condition
Administrator TOOTAI
- [asterisk-users] AMI: anything to glue originate to events?
c.savinovich at itntelecom.com
- [asterisk-users] Determine negotiated codec in script
Tom Browning
- [asterisk-users] 10.0.0-rc1: dahdi doesn't see card
Andrew Thomas
- [asterisk-users] Call files and spool directiory shared amongst several asterisk servers
Ishfaq Malik
- [asterisk-users] Polycom Phantom Ringing
eherr
- [asterisk-users] Setting outbound PRI Callerid with Asterisk 10.0.0-beta2
asterisk users
- [asterisk-users] Queue: The call keep going to agent until the agent drop the call
bilal ghayyad
- [asterisk-users] How to use password file with Authenticate Application
virendra bhati
- [asterisk-users] Dependencies for BETTER_BACKTRACES on Centos 5.6
Ishfaq Malik
- [asterisk-users] video calls not working
virendra bhati
- [asterisk-users] vigor 2920 problems
Arthur Stanfield
- [asterisk-users] CDR mysql with asterisk 1.4
salaheddine elharit
- [asterisk-users] queue ring delay
Douglas Mortensen
- [asterisk-users] AEX410P drops DTMF digits
Danny Nicholas
- [asterisk-users] no sound with ICES ?
listes at thomasi.be
- [asterisk-users] Resell VoIP Servcies
Jai Rangi
- [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
Jonas Kellens
- [asterisk-users] sip show peers
Jerry Geis
- [asterisk-users] (no subject)
Charles Wang
- [asterisk-users] How to program a 100ms delay between the ringing of queued calls w/ ringall
Douglas Mortensen
- [asterisk-users] Sending more than one call the agent while he is already in a call !! ringinuse=no/yes
bilal ghayyad
- [asterisk-users] DONT_OPTIMISE, BETTER_BACKTRACES and performance
Ishfaq Malik
- [asterisk-users] Is it possible call land into extensions.ael configuration file not in extensions.conf
virendra bhati
- [asterisk-users] MWI for non-subscribed Realtime peers?
Jan Blom
- [asterisk-users] Is it possible call land into extensions.ael configuration file not in extensions.conf
virendra bhati
- [asterisk-users] PSTN Frequency parameters
Gopalakrishnan N
- [asterisk-users] Rgarding asterisk 10 stable release
Deka, Rajib IN MAA SL
- [asterisk-users] Sporadic yellow alarms in dahdi_tool output
Ishwar Sridharan
- [asterisk-users] hwo to stok variable wiith menu
salaheddine elharit
- [asterisk-users] android won't play wav49: how to change format
sean darcy
- [asterisk-users] A new hack?
Gordon Henderson
- [asterisk-users] Displaying entered digits in the LCD of the IP Phone when is requested to enter it
bilal ghayyad
- [asterisk-users] centos asterisk 1.8 rpms: chan_gtalk and res_jabber missing?
Gaurav P
- [asterisk-users] Queue-Tip/Adhearsion installation tip
Olivier
- [asterisk-users] AsteriskGUI - Which version for 1.8 ?
Olivier
- [asterisk-users] Call Parking Realtime
Bryant Zimmerman
- [asterisk-users] Call Parking Realtime
Bryant Zimmerman
- [asterisk-users] app_mysql and asterisk 1.4
salaheddine elharit
- [asterisk-users] SIM to E1 gateway, and SMS gateway
bilal ghayyad
- [asterisk-users] When dialing the number, I need to see it in the Cisco LCD Phone
bilal ghayyad
- [asterisk-users] When dialing the number, I need to see it in the Cisco LCD Phone
Bryant Zimmerman
- [asterisk-users] When dialing the number, I need to see it in the Cisco LCD Phone
bilal ghayyad
- [asterisk-users] Best VoIP conferencing phone ?
virendra bhati
- [asterisk-users] s/n ratio detection etc...
Yasin SULUHAN
- [asterisk-users] Question on PAP2 linksys showing off-hook
Jerry Geis
- [asterisk-users] vall directly extensions from E1-PRI line
dsidir at hcmr.gr
- [asterisk-users] Sound files with MixMonitor not playable with Media Player
Jonas Kellens
- [asterisk-users] Walkie talkie to sip phone interface
Ferdinand Babas
- [asterisk-users] Walkie talkie to sip phone interfacere:
Dave Platt
Last message date:
Wed Nov 30 22:46:14 CST 2011
Archived on: Wed Nov 30 22:49:49 CST 2011
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